draft-ietf-taps-transports-08.txt   draft-ietf-taps-transports-09.txt 
Network Working Group G. Fairhurst, Ed. Network Working Group G. Fairhurst, Ed.
Internet-Draft University of Aberdeen Internet-Draft University of Aberdeen
Intended status: Informational B. Trammell, Ed. Intended status: Informational B. Trammell, Ed.
Expires: June 10, 2016 M. Kuehlewind, Ed. Expires: July 31, 2016 M. Kuehlewind, Ed.
ETH Zurich ETH Zurich
December 08, 2015 January 28, 2016
Services provided by IETF transport protocols and congestion control Services provided by IETF transport protocols and congestion control
mechanisms mechanisms
draft-ietf-taps-transports-08 draft-ietf-taps-transports-09
Abstract Abstract
This document describes transport services provided by existing IETF This document describes, surveys, classifies and compares the
protocols. It is designed to help application and network stack protocol mechanisms provided by existing IETF protocols, as
programmers and to inform the work of the IETF TAPS Working Group. background for determining a common set of transport services. It
examines the Transmission Control Protocol (TCP), Multipath TCP, the
Stream Control Transmission Protocol (SCTP), the User Datagram
Protocol (UDP), UDP-Lite, the Datagram Congestion Control Protocol
(DCCP), the Internet Control Message Protocol (ICMP), the Realtime
Transport Protocol (RTP), File Delivery over Unidirectional
Transport/Asynchronous Layered Coding Reliable Multicast (FLUTE/ALC),
and NACK-Oriented Reliable Multicast (NORM), Transport Layer Security
(TLS), Datagram TLS (DTLS), and the Hypertext Transport Protocol
(HTTP) when used as a pseudotransport.
Status of This Memo Status of This Memo
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provisions of BCP 78 and BCP 79. provisions of BCP 78 and BCP 79.
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This Internet-Draft will expire on June 10, 2016. This Internet-Draft will expire on July 31, 2016.
Copyright Notice Copyright Notice
Copyright (c) 2015 IETF Trust and the persons identified as the Copyright (c) 2016 IETF Trust and the persons identified as the
document authors. All rights reserved. document authors. All rights reserved.
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Table of Contents Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 3 1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 3
2. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 4 1.1. Overview of Transport Features . . . . . . . . . . . . . 4
3. Transport Service Features . . . . . . . . . . . . . . . . . 4 2. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 5
3.1. Congestion Control . . . . . . . . . . . . . . . . . . . 5 3. Existing Transport Protocols . . . . . . . . . . . . . . . . 5
4. Existing Transport Protocols . . . . . . . . . . . . . . . . 6 3.1. Transport Control Protocol (TCP) . . . . . . . . . . . . 6
4.1. Transport Control Protocol (TCP) . . . . . . . . . . . . 6 3.1.1. Protocol Description . . . . . . . . . . . . . . . . 6
4.1.1. Protocol Description . . . . . . . . . . . . . . . . 6 3.1.2. Interface description . . . . . . . . . . . . . . . . 8
4.1.2. Interface description . . . . . . . . . . . . . . . . 8 3.1.3. Transport Features . . . . . . . . . . . . . . . . . 8
4.1.3. Transport Features . . . . . . . . . . . . . . . . . 8 3.2. Multipath TCP (MPTCP) . . . . . . . . . . . . . . . . . . 9
4.2. Multipath TCP (MPTCP) . . . . . . . . . . . . . . . . . . 9 3.2.1. Protocol Description . . . . . . . . . . . . . . . . 9
4.2.1. Protocol Description . . . . . . . . . . . . . . . . 9 3.2.2. Interface Description . . . . . . . . . . . . . . . . 9
4.2.2. Interface Description . . . . . . . . . . . . . . . . 9 3.2.3. Transport features . . . . . . . . . . . . . . . . . 10
4.2.3. Transport features . . . . . . . . . . . . . . . . . 10 3.3. Stream Control Transmission Protocol (SCTP) . . . . . . . 10
4.3. Stream Control Transmission Protocol (SCTP) . . . . . . . 10 3.3.1. Protocol Description . . . . . . . . . . . . . . . . 11
4.3.1. Protocol Description . . . . . . . . . . . . . . . . 11 3.3.2. Interface Description . . . . . . . . . . . . . . . . 13
4.3.2. Interface Description . . . . . . . . . . . . . . . . 13 3.3.3. Transport Features . . . . . . . . . . . . . . . . . 15
4.3.3. Transport Features . . . . . . . . . . . . . . . . . 15 3.4. User Datagram Protocol (UDP) . . . . . . . . . . . . . . 16
4.4. User Datagram Protocol (UDP) . . . . . . . . . . . . . . 16 3.4.1. Protocol Description . . . . . . . . . . . . . . . . 16
4.4.1. Protocol Description . . . . . . . . . . . . . . . . 16 3.4.2. Interface Description . . . . . . . . . . . . . . . . 17
4.4.2. Interface Description . . . . . . . . . . . . . . . . 17 3.4.3. Transport Features . . . . . . . . . . . . . . . . . 17
4.4.3. Transport Features . . . . . . . . . . . . . . . . . 18 3.5. Lightweight User Datagram Protocol (UDP-Lite) . . . . . . 18
4.5. Lightweight User Datagram Protocol (UDP-Lite) . . . . . . 18 3.5.1. Protocol Description . . . . . . . . . . . . . . . . 18
4.5.1. Protocol Description . . . . . . . . . . . . . . . . 18 3.5.2. Interface Description . . . . . . . . . . . . . . . . 19
4.5.2. Interface Description . . . . . . . . . . . . . . . . 19 3.5.3. Transport Features . . . . . . . . . . . . . . . . . 19
4.5.3. Transport Features . . . . . . . . . . . . . . . . . 19 3.6. Datagram Congestion Control Protocol (DCCP) . . . . . . . 19
4.6. Datagram Congestion Control Protocol (DCCP) . . . . . . . 20 3.6.1. Protocol Description . . . . . . . . . . . . . . . . 20
4.6.1. Protocol Description . . . . . . . . . . . . . . . . 20 3.6.2. Interface Description . . . . . . . . . . . . . . . . 21
4.6.2. Interface Description . . . . . . . . . . . . . . . . 21 3.6.3. Transport Features . . . . . . . . . . . . . . . . . 21
4.6.3. Transport Features . . . . . . . . . . . . . . . . . 22 3.7. Internet Control Message Protocol (ICMP) . . . . . . . . 22
4.7. Internet Control Message Protocol (ICMP) . . . . . . . . 22 3.7.1. Protocol Description . . . . . . . . . . . . . . . . 22
4.7.1. Protocol Description . . . . . . . . . . . . . . . . 23 3.7.2. Interface Description . . . . . . . . . . . . . . . . 23
4.7.2. Interface Description . . . . . . . . . . . . . . . . 24 3.7.3. Transport Features . . . . . . . . . . . . . . . . . 23
4.7.3. Transport Features . . . . . . . . . . . . . . . . . 24 3.8. Realtime Transport Protocol (RTP) . . . . . . . . . . . . 23
4.8. Realtime Transport Protocol (RTP) . . . . . . . . . . . . 24 3.8.1. Protocol Description . . . . . . . . . . . . . . . . 24
4.8.1. Protocol Description . . . . . . . . . . . . . . . . 24 3.8.2. Interface Description . . . . . . . . . . . . . . . . 25
4.8.2. Interface Description . . . . . . . . . . . . . . . . 25 3.8.3. Transport Features . . . . . . . . . . . . . . . . . 25
4.8.3. Transport Features . . . . . . . . . . . . . . . . . 26 3.9. File Delivery over Unidirectional Transport/Asynchronous
4.9. File Delivery over Unidirectional Transport/Asynchronous Layered Coding Reliable Multicast (FLUTE/ALC) . . . . . . 25
Layered Coding Reliable Multicast (FLUTE/ALC) . . . . . . 26 3.9.1. Protocol Description . . . . . . . . . . . . . . . . 26
4.9.1. Protocol Description . . . . . . . . . . . . . . . . 27 3.9.2. Interface Description . . . . . . . . . . . . . . . . 28
4.9.2. Interface Description . . . . . . . . . . . . . . . . 29 3.9.3. Transport Features . . . . . . . . . . . . . . . . . 28
4.9.3. Transport Features . . . . . . . . . . . . . . . . . 29 3.10. NACK-Oriented Reliable Multicast (NORM) . . . . . . . . . 29
4.10. NACK-Oriented Reliable Multicast (NORM) . . . . . . . . . 30 3.10.1. Protocol Description . . . . . . . . . . . . . . . . 29
4.10.1. Protocol Description . . . . . . . . . . . . . . . . 30 3.10.2. Interface Description . . . . . . . . . . . . . . . 30
4.10.2. Interface Description . . . . . . . . . . . . . . . 31 3.10.3. Transport Features . . . . . . . . . . . . . . . . . 30
4.10.3. Transport Features . . . . . . . . . . . . . . . . . 31 3.11. Transport Layer Security (TLS) and Datagram TLS (DTLS) as
4.11. Transport Layer Security (TLS) and Datagram TLS (DTLS) as a pseudotransport . . . . . . . . . . . . . . . . . . . . 31
a pseudotransport . . . . . . . . . . . . . . . . . . . . 32 3.11.1. Protocol Description . . . . . . . . . . . . . . . . 31
4.11.1. Protocol Description . . . . . . . . . . . . . . . . 32 3.11.2. Interface Description . . . . . . . . . . . . . . . 32
4.11.2. Interface Description . . . . . . . . . . . . . . . 33 3.11.3. Transport Features . . . . . . . . . . . . . . . . . 33
4.11.3. Transport Features . . . . . . . . . . . . . . . . . 34 3.12. Hypertext Transport Protocol (HTTP) over TCP as a
4.12. Hypertext Transport Protocol (HTTP) over TCP as a pseudotransport . . . . . . . . . . . . . . . . . . . . . 34
pseudotransport . . . . . . . . . . . . . . . . . . . . . 35 3.12.1. Protocol Description . . . . . . . . . . . . . . . . 35
4.12.1. Protocol Description . . . . . . . . . . . . . . . . 35 3.12.2. Interface Description . . . . . . . . . . . . . . . 35
4.12.2. Interface Description . . . . . . . . . . . . . . . 36 3.12.3. Transport features . . . . . . . . . . . . . . . . . 36
4.12.3. Transport features . . . . . . . . . . . . . . . . . 37 4. Congestion Control . . . . . . . . . . . . . . . . . . . . . 37
5. Transport Service Features . . . . . . . . . . . . . . . . . 37 5. Transport Features . . . . . . . . . . . . . . . . . . . . . 38
6. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 41 6. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 42
7. Security Considerations . . . . . . . . . . . . . . . . . . . 41 7. Security Considerations . . . . . . . . . . . . . . . . . . . 42
8. Contributors . . . . . . . . . . . . . . . . . . . . . . . . 41 8. Contributors . . . . . . . . . . . . . . . . . . . . . . . . 42
9. Acknowledgments . . . . . . . . . . . . . . . . . . . . . . . 42 9. Acknowledgments . . . . . . . . . . . . . . . . . . . . . . . 43
10. Informative References . . . . . . . . . . . . . . . . . . . 42 10. Informative References . . . . . . . . . . . . . . . . . . . 43
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 52 Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 53
1. Introduction 1. Introduction
Internet applications make use of the Services provided by a Internet applications make use of the Services provided by a
Transport protocol, such as TCP (a reliable, in-order stream Transport protocol, such as TCP (a reliable, in-order stream
protocol) or UDP (an unreliable datagram protocol). We use the term protocol) or UDP (an unreliable datagram protocol). We use the term
"Transport Service" to mean the end-to-end service provided to an "Transport Service" to mean the end-to-end service provided to an
application by the transport layer. That service can only be application by the transport layer. That service can only be
provided correctly if information about the intended usage is provided correctly if information about the intended usage is
supplied from the application. The application may determine this supplied from the application. The application may determine this
information at design time, compile time, or run time, and may information at design time, compile time, or run time, and may
include guidance on whether a feature is required, a preference by include guidance on whether a feature is required, a preference by
the application, or something in between. Examples of features of the application, or something in between. Examples of features of
Transport Services are reliable delivery, ordered delivery, content Transport Services are reliable delivery, ordered delivery, content
privacy to in-path devices, and integrity protection. privacy to in-path devices, and integrity protection.
The IETF has defined a wide variety of transport protocols beyond TCP The IETF has defined a wide variety of transport protocols beyond TCP
and UDP, including SCTP, DCCP, MP-TCP, and UDP-Lite. Transport and UDP, including SCTP, DCCP, MPTCP, and UDP-Lite. Transport
services may be provided directly by these transport protocols, or services may be provided directly by these transport protocols, or
layered on top of them using protocols such as WebSockets (which runs layered on top of them using protocols such as WebSockets (which runs
over TCP), RTP (over TCP or UDP) or WebRTC data channels (which run over TCP), RTP (over TCP or UDP) or WebRTC data channels (which run
over SCTP over DTLS over UDP or TCP). Services built on top of UDP over SCTP over DTLS over UDP or TCP). Services built on top of UDP
or UDP-Lite typically also need to specify additional mechanisms, or UDP-Lite typically also need to specify additional mechanisms,
including a congestion control mechanism (such as NewReno, TFRC or including a congestion control mechanism (such as NewReno, TFRC or
LEDBAT). This extends the set of available Transport Services beyond LEDBAT). This extends the set of available Transport Services beyond
those provided to applications by TCP and UDP. those provided to applications by TCP and UDP.
1.1. Overview of Transport Features
Transport protocols can be differentiated by the features of the
services they provide.
Some of these provided features are closely related to basic control
function that a protocol needs to work over a network path, such as
addressing. The number of participants in a given association also
determines its applicability: if a connection is between endpoints
(unicast), to one of multiple endpoints (anycast), and/or
simultaneously to multiple endpoints (multicast). Unicast protocols
usually support bidirectional communication, while multicast is
generally unidirectional. Another feature is whether a transport
requires a control exchange across the network at setup (e.g., TCP),
or whether it connection-less (e.g., UDP).
For the delivery of the packets itself, reliability and integrity
protection, ordering, and framing are basic features. However, these
features are implemented with different levels of assurance in
different protocols. As an example, a transport service may provide
full reliability, providing detection of loss and retransmission
(e.g., TCP). SCTP offers a message-based service that can provide
full or partial reliability, and allows the protocol to minimize the
head of line blocking due to the support of ordered and unordered
message delivery within multiple streams. UDP-Lite and DCCP can
provide partial integrity protection to enable corruption tolerance.
Usually a protocol has been designed to support one specific type of
delivery/framing: data either needs to be divided into transmission
units based on network packets (datagram service), a data stream is
segmented and re-combined across multiple packets (stream service),
or whole objects such as files are handled accordingly. This
decision strongly influences the interface that is provided to the
upper layer.
In addition, transport protocols offer a certain support on
transmission control. For example, a transport service can provide
flow control to allow a receiver to regulate the transmission rate of
a sender. Further a transport service can provide congestion control
(see Section 4). As an example TCP and SCTP provide congestion
control for use in the Internet, whereas UDP leaves this function to
the upper layer protocol that uses UDP.
Security features are often provided independent of the transport
protocol, via Transport Layer Security (TLS, see {{transport-layer-
security-tls-and- datagram-tls-dtls-as-a-pseudotransport}}) or by the
application layer protocol itself. The security properties TLS
provides to the application (such as confidentiality, integrity, and
authenticity) are also features of the transport layer, even though
they are often presently implemented in a separate protocol.
2. Terminology 2. Terminology
The following terms are defined throughout this document, and in The following terms are used throughout this document, and in
subsequent documents produced by TAPS describing the composition and subsequent documents produced by TAPS that describe the composition
decomposition of transport services. and decomposition of transport services.
Transport Service Feature: a specific end-to-end feature that a Transport Service Feature: a specific end-to-end feature that the
transport service provides to its clients. Examples include transport layer provides to an application. Examples include
confidentiality, reliable delivery, ordered delivery, message- confidentiality, reliable delivery, ordered delivery, message-
versus-stream orientation, etc. versus-stream orientation, etc.
Transport Service: a set of transport service features, without an Transport Service: a set of Transport Features, without an
association to any given framing protocol, which provides a association to any given framing protocol, which provides a
complete service to an application. complete service to an application.
Transport Protocol: an implementation that provides one or more Transport Protocol: an implementation that provides one or more
different transport services using a specific framing and header different transport services using a specific framing and header
format on the wire. format on the wire.
Transport Protocol Component: an implementation of a transport
service feature within a protocol.
Transport Service Instance: an arrangement of transport protocols Transport Service Instance: an arrangement of transport protocols
with a selected set of features and configuration parameters that with a selected set of features and configuration parameters that
implements a single transport service, e.g., a protocol stack (RTP implements a single transport service, e.g., a protocol stack (RTP
over UDP). over UDP).
Application: an entity that uses the transport layer for end-to-end Application: an entity that uses the transport layer for end-to-end
delivery data across the network (this may also be an upper layer delivery data across the network (this may also be an upper layer
protocol or tunnel encapsulation). protocol or tunnel encapsulation).
3. Transport Service Features 3. Existing Transport Protocols
Transport protocols can be differentiated by the features of the
services they provide.
One fundamental feature is whether a transport offers a service that
divides the data into transmission units based on network packets
(known as a Datagram service), or whether it combines and segments
data across multiple packets (e.g., the Stream service provided by
TCP).
Another fundamental feature is whether a transport requires a control
exchange across the network at setup (e.g., TCP), or whether it
connection-less (e.g., UDP).
A transport service can also offer reliability, for instance, SCTP
offers a message-based service providing full or partial reliability
and allowing to minimize the head of line blocking due to the support
of unordered and unordered message delivery within multiple streams,
UDP-Lite and DCCP provide partial integrity protection.
A transport service can provide congestion control (see Section 3.1).
TCP and SCTP provide congestion control for use in the Internet,
whereas UDP leaves this function to the upper layer protocol that
uses UDP. DCCP offers a range of congestion control approaches and
LEDBAT can support low-priority "scavenger" communication, intending
to defer use of capacity to other Internet flows sharing a congested
bottleneck.
Transport services may be unidirectional or bidirectional, to a
single a single endpoint, to one of multiple endpoints, or multicast
simultaneously to multiple endpoints.
The service offered by transport protocols and frameworks can also be
differentiated in many other ways.
3.1. Congestion Control
Congestion control is critical to the stable operation of the
Internet, applications and other protocols that choose to use a
datagram protocol (e.g., UDP or UDP-Lite) need to employ mechanisms
to prevent congestion collapse and to establish some degree of
fairness with concurrent traffic.
A variety of techniques are used to provide congestion control in the
Internet. Each technique requires that the protocol provide a method
for deriving the metric the congestion control algorithm uses to
detect congestion and the property of a packet it uses to determine
when to send. Given these relatively wide constraints, the
congestion control techniques that can be applied by different
transport protocols are largely orthogonal to the choice of transport
protocols themselves. This section provides an overview of the
congestion control techniques available to the protocols described in
Section 4.
Most commonly deployed congestion control mechanisms use one of three
mechanisms to detect congestion:
o detection of loss, which is interpreted as a congestion signal;
o Explicit Congestion Notification (ECN) [RFC3168] to provide
explicit signaling of congestion without inducing loss (see
[I-D.ietf-aqm-ecn-benefits]); and/or
o a retransmission timer with exponential back-off.
Protocols such as SCTP and TCP [RFC5681] that use sliding-window-
based receiver flow control commonly use a separate congestion window
for congestion control. Each time congestion is detected, this
separate congestion window is reduced. Data in flight is capped to
the minimum of the two windows. This approach is also used by DCCP
CCID-2 for datagram congestion control.
Rate-based methods have also been defined based on the loss ratio and
observed round trip time, such as TFRC [RFC5348] and TFRC-SP
[RFC4828]. These methods utlise a throughput equation to determine
the maximum acceptable rate. Such methods are used with DCCP CCID-3
[RFC4342] and CCID-4 [RFC5622], WEBRC [RFC3738], and other
applications.
In addition, a congestion control mechanism may react to changes in
delay as an indication for congestion. Delay-based congestion
detection methods tend to induce less loss than loss-based methods,
and therefore generally do not compete well with them across shared
bottleneck links. However, such methods, such as LEDBAT [RFC6824],
are are deployed in the Internet for scavenger traffic, which will
use unused capacity but readily yield to presumably interactive or
otherwise higher-priority, loss-based congestion-controlled traffic.
4. Existing Transport Protocols
This section provides a list of known IETF transport protocols and This section provides a list of known IETF transport protocols and
transport protocol frameworks. It does not make an assessment about transport protocol frameworks. It does not make an assessment about
whether specific implementations of protocols are fully compliant to whether specific implementations of protocols are fully compliant to
current IETF specifications. current IETF specifications.
4.1. Transport Control Protocol (TCP) 3.1. Transport Control Protocol (TCP)
TCP is an IETF standards track transport protocol. [RFC0793] TCP is an IETF standards track transport protocol. [RFC0793]
introduces TCP as follows: "The Transmission Control Protocol (TCP) introduces TCP as follows: "The Transmission Control Protocol (TCP)
is intended for use as a highly reliable host-to-host protocol is intended for use as a highly reliable host-to-host protocol
between hosts in packet-switched computer communication networks, and between hosts in packet-switched computer communication networks, and
in interconnected systems of such networks." Since its introduction, in interconnected systems of such networks." Since its introduction,
TCP has become the default connection- oriented, stream-based TCP has become the default connection- oriented, stream-based
transport protocol in the Internet. It is widely implemented by transport protocol in the Internet. It is widely implemented by
endpoints and widely used by common applications. endpoints and widely used by common applications.
4.1.1. Protocol Description 3.1.1. Protocol Description
TCP is a connection-oriented protocol, providing a three way TCP is a connection-oriented protocol, providing a three way
handshake to allow a client and server to set up a connection and handshake to allow a client and server to set up a connection and
negotiate features, and mechanisms for orderly completion and negotiate features, and mechanisms for orderly completion and
immediate teardown of a connection. TCP is defined by a family of immediate teardown of a connection. TCP is defined by a family of
RFCs [RFC4614]. RFCs [RFC7414].
TCP provides multiplexing to multiple sockets on each host using port TCP provides multiplexing to multiple sockets on each host using port
numbers. A similar approach is adopted by other IETF-defined numbers. A similar approach is adopted by other IETF-defined
transports. An active TCP session is identified by its four-tuple of transports. An active TCP session is identified by its four-tuple of
local and remote IP addresses and local port and remote port numbers. local and remote IP addresses and local port and remote port numbers.
The destination port during connection setup is often used to The destination port during connection setup is often used to
indicate the requested service. indicate the requested service.
TCP partitions a continuous stream of bytes into segments, sized to TCP partitions a continuous stream of bytes into segments, sized to
fit in IP packets. ICMP-based Path MTU discovery [RFC1191][RFC1981] fit in IP packets based on a negotiated maximum segment size and
as well as Packetization Layer Path MTU Discovery (PMTUD) [RFC4821] further constrained by the effective MTU from PMTUD. ICMP-based Path
have been defined by the IETF. MTU discovery [RFC1191][RFC1981] as well as Packetization Layer Path
MTU Discovery (PMTUD) [RFC4821] have been defined by the IETF.
Each byte in the stream is identified by a sequence number. The Each byte in the stream is identified by a sequence number. The
sequence number is used to order segments on receipt, to identify sequence number is used to order segments on receipt, to identify
segments in acknowledgments, and to detect unacknowledged segments segments in acknowledgments, and to detect unacknowledged segments
for retransmission. This is the basis of the reliable, ordered for retransmission. This is the basis of the reliable, ordered
delivery of data in a TCP stream. TCP Selective Acknowledgment delivery of data in a TCP stream. TCP Selective Acknowledgment
[RFC2018] extends this mechanism by making it possible to identify (SACK) [RFC2018] extends this mechanism by making it possible to
missing segments more precisely, reducing spurious retransmission. provide earlier identification of which segments are missing,
allowing faster retransmission. SACK-based methods (e.g. DSACK) can
also result in less spurious retransmission.
Receiver flow control is provided by a sliding window: limiting the Receiver flow control is provided by a sliding window: limiting the
amount of unacknowledged data that can be outstanding at a given amount of unacknowledged data that can be outstanding at a given
time. The window scale option [RFC7323] allows a receiver to use time. The window scale option [RFC7323] allows a receiver to use
windows greater than 64KB. windows greater than 64KB.
TCP provides congestion control [RFC5681], described further in All TCP senders provide congestion control, such as described in
Section 3.1 below. [RFC5681]. TCP's congestion control mechanism is tied to a sliding
window as well [RFC5681]. Examples for different kind of congestion
control schemes are given in Section 4. The sending window at a
given point in time is the minimum of the receiver window and the
congestion window. The congestion window is increased in the absence
of congestion and, respectively, decreased if congestion is detected.
Often loss is implicitly handled as a congestion indication which is
detected in TCP (also as input for retransmission handling) based on
two mechanisms: A retransmission timer with exponential back-up or
the reception of three acknowledgment for the same segment, so called
duplicated ACKs (Fast retransmit). In addition, Explicit Congestion
Notification (ECN) [RFC3168] can be used in TCP, if supported by both
endpoints, that allows a network node to signal congestion without
inducing loss. Alternatively, a delay-based congestion control
scheme can be used in TCP that reacts to changes in delay as an early
indication of congestion as also further described in Section 4.
TCP protocol instances can be extended [RFC4614] and tuned. Some TCP protocol instances can be extended [RFC7414] and tuned. Some
features are sender-side only, requiring no negotiation with the features are sender-side only, requiring no negotiation with the
receiver; some are receiver-side only, some are explicitly negotiated receiver; some are receiver-side only, some are explicitly negotiated
during connection setup. during connection setup.
By default, TCP segment partitioning uses Nagle's algorithm [RFC0896] TCP may buffer data, e.g., to optimize processing or capacity usage.
to buffer data at the sender into large segments, potentially TCP can therefore provides mechanisms to control this, including an
incurring sender-side buffering delay; this algorithm can be disabled optional "PUSH" function [RFC0793] that explicitly requests the
by the sender to transmit more immediately, e.g., to reduce latency transport service not to delay data. By default, TCP segment
for interactive sessions. partitioning uses Nagle's algorithm [RFC0896] to buffer data at the
sender into large segments, potentially incurring sender-side
buffering delay; this algorithm can be disabled by the sender to
transmit more immediately, e.g., to reduce latency for interactive
sessions.
TCP provides an "urgent data" function for limited out-of-order TCP provides an "urgent data" function for limited out-of-order
delivery of the data. This function is deprecated [RFC6093]. delivery of the data. This function is deprecated [RFC6093].
A mandatory checksum provides a basic integrity check against A mandatory checksum provides a basic integrity check against
misdelivery and data corruption over the entire packet. Applications misdelivery and data corruption over the entire packet. Applications
that require end to end integrity of data are recommended to include that require end to end integrity of data are recommended to include
a stronger integrity check of their payload data. The TCP checksum a stronger integrity check of their payload data. The TCP checksum
does not support partial corruption protection (as in DCCP/UDP-Lite). does not support partial payload protection (as in DCCP/UDP-Lite).
TCP supports only unicast connections. TCP supports only unicast connections.
4.1.2. Interface description 3.1.2. Interface description
A User/TCP Interface is defined in [RFC0793] providing six user A User/TCP Interface is defined in [RFC0793] providing six user
commands: Open, Send, Receive, Close, Status. This interface does commands: Open, Send, Receive, Close, Status. This interface does
not describe configuration of TCP options or parameters beside use of not describe configuration of TCP options or parameters beside use of
the PUSH and URGENT flags. the PUSH and URGENT flags.
[RFC1122] describes extensions of the TCP/application layer interface [RFC1122] describes extensions of the TCP/application layer interface
for: for:
o reporting soft errors such as reception of ICMP error messages, o reporting soft errors such as reception of ICMP error messages,
extensive retransmission or urgent pointer advance, extensive retransmission or urgent pointer advance,
o providing a possibility to specify the Differentiated Services o providing a possibility to specify the Differentiated Services
Code Point (DSCP) (formerly, the Type-of-Service, TOS) for Code Point (DSCP) [RFC3260] (formerly, the Type-of-Service, TOS)
segments, for segments,
o providing a flush call to empty the TCP send queue, and o providing a flush call to empty the TCP send queue, and
o multihoming support. o multihoming support.
In API implementations derived from the BSD Sockets API, TCP sockets In API implementations derived from the BSD Sockets API, TCP sockets
are created using the "SOCK_STREAM" socket type as described in the are created using the "SOCK_STREAM" socket type as described in the
IEEE Portable Operating System Interface (POSIX) Base Specifications IEEE Portable Operating System Interface (POSIX) Base Specifications
[POSIX]. The features used by a protocol instance may be set and [POSIX]. The features used by a protocol instance may be set and
tuned via this API. There are current no documents in the RFC Series tuned via this API. There are currently no documents in the RFC
that describe this interface. Series that describe this interface.
4.1.3. Transport Features 3.1.3. Transport Features
The transport features provided by TCP are: The transport features provided by TCP are:
o unicast transport o connection-oriented transport with feature negotiation and
application-to-port mapping (implemented using SYN segments and
the TCP option field to negotiate features),
o connection setup with feature negotiation and application-to-port o unicast transport (though anycast TCP is implemented, at risk of
mapping, implemented using SYN segments and the TCP option field instability due to rerouting),
to negotiate features.
o port multiplexing: each TCP session is uniquely identified by a o port multiplexing,
combination of the ports and IP address fields.
o Uni-or bidirectional communication. o uni- or bidirectional communication,
o stream-oriented delivery in a single stream. o stream-oriented delivery in a single stream,
o fully reliable delivery, implemented using ACKs sent from the o fully reliable delivery (implemented using ACKs sent from the
receiver to confirm delivery. receiver to confirm delivery),
o error detection: a segment checksum verifies delivery to the o error detection (implemented using a segment checksum to verify
correct endpoint and integrity of the data and options. delivery to the correct endpoint and integrity of the data and
options),
o segmentation: packets are fragmented to a negotiated maximum o segmentation,
segment size, further constrained by the effective MTU from PMTUD.
o data bundling, an optional mechanism that uses Nagle's algorithm o data bundling (optional; uses Nagle's algorithm to coalesce data
to coalesce data sent within the same RTT into full-sized sent within the same RTT into full-sized segments),
segments.
o flow control using a window-based mechanism, where the receiver o flow control (implemented using a window-based mechanism where the
advertises the window that it is willing to buffer. receiver advertises the window that it is willing to buffer),
o congestion control: a window-based method that uses Additive o congestion control (usually implemented using a window-based
Increase Multiplicative Decrease (AIMD) to control the sending mechanism and four algorithm for different phases of the
rate and to conservatively choose a rate after congestion is transmission: slow start, congestion avoidance, fast retransmit,
detected. and fast recovery [RFC5681]).
4.2. Multipath TCP (MPTCP) 3.2. Multipath TCP (MPTCP)
Multipath TCP [RFC6824] is an extension for TCP to support multi- Multipath TCP [RFC6824] is an extension for TCP to support multi-
homing. It is designed to be as transparent as possible to middle- homing for resilience, mobility and load-balancing. It is designed
boxes. It does so by establishing regular TCP flows between a pair to be as transparent as possible to middleboxes. It does so by
of source/destination endpoints, and multiplexing the application's establishing regular TCP flows between a pair of source/destination
stream over these flows. endpoints, and multiplexing the application's stream over these
flows. Sub-flows can be started over IPv4 or IPv6 for the same
session.
4.2.1. Protocol Description 3.2.1. Protocol Description
MPTCP uses TCP options for its control plane. They are used to MPTCP uses TCP options for its control plane. They are used to
signal multipath capabilities, as well as to negotiate data sequence signal multipath capabilities, as well as to negotiate data sequence
numbers, and advertise other available IP addresses and establish new numbers, and advertise other available IP addresses and establish new
sessions between pairs of endpoints. sessions between pairs of endpoints.
4.2.2. Interface Description By multiplexing one byte stream over separate paths, MPTCP can
achieve a higher throughput than TCP in certain situations. However,
if coupled congestion control [RFC6356] is used, it might limit this
benefit to maintain fairness to other flows at the bottleneck. When
aggregating capacity over multiple paths, and depending on the way
packets are scheduled on each TCP subflow, additional delay and
higher jitter might be observed observed before in-order delivery of
data to the applications.
3.2.2. Interface Description
By default, MPTCP exposes the same interface as TCP to the By default, MPTCP exposes the same interface as TCP to the
application. [RFC6897] however describes a richer API for MPTCP- application. [RFC6897] however describes a richer API for MPTCP-
aware applications. aware applications.
This Basic API describes how an application can: This Basic API describes how an application can:
o enable or disable MPTCP. o enable or disable MPTCP.
o bind a socket to one or more selected local endpoints. o bind a socket to one or more selected local endpoints.
o query local and remote endpoint addresses. o query local and remote endpoint addresses.
o get a unique connection identifier (similar to an address-port o get a unique connection identifier (similar to an address-port
pair for TCP). pair for TCP).
The document also recommends the use of extensions defined for SCTP The document also recommends the use of extensions defined for SCTP
[RFC6458] (see next section) to support multihoming. [RFC6458] (see next section) to support multihoming for resilience
and mobility.
4.2.3. Transport features 3.2.3. Transport features
As an extension to TCP, MPTCP provides mostly the same features. By As an extension to TCP, MPTCP provides mostly the same features. By
establishing multiple sessions between available endpoints, it can establishing multiple sessions between available endpoints, it can
additionally provide soft failover solutions should one of the paths additionally provide soft failover solutions should one of the paths
become unusable. In addition, by multiplexing one byte stream over become unusable.
separate paths, it can achieve a higher throughput than TCP in
certain situations. Note, however, that coupled congestion control
[RFC6356] might limit this benefit to maintain fairness to other
flows at the bottleneck. When aggregating capacity over multiple
paths, and depending on the way packets are scheduled on each TCP
subflow, an additional delay and higher jitter might be observed
observed before in-order delivery of data to the applications.
The transport features provided by MPTCP in addition to TCP therefore The transport features provided by MPTCP in addition to TCP therefore
are: are:
o congestion control with load balancing over multiple connections. o multihoming for load-balancing, with endpoint multiplexing of a
single byte stream, using either coupled congestion control or for
o endpoint multiplexing of a single byte stream (higher throughput). throughput maximization,
o address family multiplexing: sub-flows can be started over IPv4 or o address family multiplexing (using IPv4 and IPv6 for the same
IPv6 for the same session. session),
o resilience to network failure and/or handover. o resilience to network failure and/or handover.
4.3. Stream Control Transmission Protocol (SCTP) 3.3. Stream Control Transmission Protocol (SCTP)
SCTP is a message-oriented IETF standards track transport protocol. SCTP is a message-oriented IETF standards track transport protocol.
The base protocol is specified in [RFC4960]. It supports multi- The base protocol is specified in [RFC4960]. It supports multi-
homing and path failover to provide resilience to path failures. An homing and path failover to provide resilience to path failures. An
SCTP association has multiple streams in each direction, providing SCTP association has multiple streams in each direction, providing
in-sequence delivery of user messages within each stream. This in-sequence delivery of user messages within each stream. This
allows it to minimize head of line blocking. SCTP supports multiple allows it to minimize head of line blocking. SCTP supports multiple
stream scheduling schemes controlling stream multiplexing, including stream scheduling schemes controlling stream multiplexing, including
priority and fair weighting schemes. priority and fair weighting schemes.
SCTP is extensible. Currently defined extensions include mechanisms SCTP was originally developed for transporting telephony signaling
for dynamic re-configuration of streams [RFC6525] and IP addresses messages and is deployed in telephony signaling networks, especially
[RFC5061]. Furthermore, the extension specified in [RFC3758]
introduces the concept of partial reliability for user messages.
SCTP was originally developed for transporting telephony signalling
messages and is deployed in telephony signalling networks, especially
in mobile telephony networks. It can also be used for other in mobile telephony networks. It can also be used for other
services, for example in the WebRTC framework for data channels. It services, for example, in the WebRTC framework for data channels.
is therefore deployed in all Web browsers supporting WebRTC.
4.3.1. Protocol Description 3.3.1. Protocol Description
SCTP is a connection-oriented protocol using a four way handshake to SCTP is a connection-oriented protocol using a four way handshake to
establish an SCTP association, and a three way message exchange to establish an SCTP association, and a three way message exchange to
gracefully shut it down. It uses the same port number concept as gracefully shut it down. It uses the same port number concept as
DCCP, TCP, UDP, and UDP-Lite. SCTP only supports unicast. DCCP, TCP, UDP, and UDP-Lite. SCTP only supports unicast.
SCTP uses the 32-bit CRC32c for protecting SCTP packets against bit SCTP uses the 32-bit CRC32c for protecting SCTP packets against bit
errors and misdelivery of packets to an unintended endpoint. This is errors and misdelivery of packets to an unintended endpoint. This is
stronger than the 16-bit checksums used by TCP or UDP. However, stronger than the 16-bit checksums used by TCP or UDP. However,
partial checksum coverage as provided by DCCP or UDP-Lite is not partial payload checksum coverage as provided by DCCP or UDP-Lite is
supported. not supported.
SCTP has been designed with extensibility in mind. Each SCTP packet SCTP has been designed with extensibility in mind. A common header
starts with a single common header containing the port numbers, a is followed by a sequence of chunks. [RFC4960] defines how a
verification tag and the CRC32c checksum. This common header is
followed by a sequence of chunks. Each chunk consists of a type
field, flags, a length field and a value. [RFC4960] defines how a
receiver processes chunks with an unknown chunk type. The support of receiver processes chunks with an unknown chunk type. The support of
extensions can be negotiated during the SCTP handshake. extensions can be negotiated during the SCTP handshake. Currently
defined extensions include mechanisms for dynamic re-configuration of
streams [RFC6525] and IP addresses [RFC5061]. Furthermore, the
extension specified in [RFC3758] introduces the concept of partial
reliability for user messages.
SCTP provides a message-oriented service. Multiple small user SCTP provides a message-oriented service. Multiple small user
messages can be bundled into a single SCTP packet to improve messages can be bundled into a single SCTP packet to improve
efficiency. For example, this bundling may be done by delaying user efficiency. For example, this bundling may be done by delaying user
messages at the sender, similar to Nagle's algorithm used by TCP. messages at the sender, similar to Nagle's algorithm used by TCP.
User messages which would result in IP packets larger than the MTU User messages which would result in IP packets larger than the MTU
will be fragmented at the sender and reassembled at the receiver. will be fragmented at the sender and reassembled at the receiver.
There is no protocol limit on the user message size. ICMP-based path There is no protocol limit on the user message size. For MTU
MTU discovery as specified for IPv4 in [RFC1191] and for IPv6 in discovery the same mechanism than for TCP can be used
[RFC1981] as well as packetization layer path MTU discovery as [RFC1981][RFC4821], as well as utilizing probe packets with padding
specified in [RFC4821] with probe packets using the padding chunks chunks, as defined in [RFC4820].
defined in [RFC4820] are supported.
[RFC4960] specifies TCP-friendly congestion control to protect the [RFC4960] specifies TCP-friendly congestion control to protect the
network against overload; see Section 3.1 for more. SCTP also uses network against overload. SCTP also uses sliding window flow control
sliding window flow control to protect receivers against overflow. to protect receivers against overflow. Similar to TCP, SCTP also
Similar to TCP, SCTP also supports delaying acknowledgments. supports delaying acknowledgments. [RFC7053] provides a way for the
[RFC7053] provides a way for the sender of user messages to request sender of user messages to request the immediate sending of the
the immediate sending of the corresponding acknowledgments. corresponding acknowledgments.
Each SCTP association has between 1 and 65536 uni-directional streams Each SCTP association has between 1 and 65536 uni-directional streams
in each direction. The number of streams can be different in each in each direction. The number of streams can be different in each
direction. Every user message is sent on a particular stream. User direction. Every user message is sent on a particular stream. User
messages can be sent un- ordered, or ordered upon request by the messages can be sent un-ordered, or ordered upon request by the upper
upper layer. Un-ordered messages can be delivered as soon as they layer. Un-ordered messages can be delivered as soon as they are
are completely received. Ordered messages sent on the same stream completely received. Ordered messages sent on the same stream are
are delivered at the receiver in the same order as sent by the delivered at the receiver in the same order as sent by the sender.
sender. For user messages not requiring fragmentation, this For user messages not requiring fragmentation, this minimizes head of
minimizes head of line blocking. line blocking.
The base protocol defined in [RFC4960] does not allow interleaving of The base protocol defined in [RFC4960] does not allow interleaving of
user- messages. Large messages on one stream can therefore block the user- messages. Large messages on one stream can therefore block the
sending of user messages on other streams. sending of user messages on other streams.
[I-D.ietf-tsvwg-sctp-ndata] overcomes this limitation. This draft [I-D.ietf-tsvwg-sctp-ndata] overcomes this limitation. This draft
also specifies multiple algorithms for the sender side selection of also specifies multiple algorithms for the sender side selection of
which streams to send data from, supporting a variety of scheduling which streams to send data from, supporting a variety of scheduling
algorithms including priority based methods. The stream re- algorithms including priority based methods. The stream re-
configuration extension defined in [RFC6525] allows streams to be configuration extension defined in [RFC6525] allows streams to be
reset during the lifetime of an association and to increase the reset during the lifetime of an association and to increase the
number of streams, if the number of streams negotiated in the SCTP number of streams, if the number of streams negotiated in the SCTP
handshake becomes insufficient. handshake becomes insufficient.
Each user message sent is either delivered to the receiver or, in Each user message sent is either delivered to the receiver or, in
case of excessive retransmissions, the association is terminated in a case of excessive retransmissions, the association is terminated in a
non-graceful way [RFC4960], similar to TCP behaviour. In addition to non-graceful way [RFC4960], similar to TCP behavior. In addition to
this reliable transfer, the partial reliability extension [RFC3758] this reliable transfer, the partial reliability extension [RFC3758]
allows a sender to abandon user messages. The application can allows a sender to abandon user messages. The application can
specify the policy for abandoning user messages. Examples of these specify the policy for abandoning user messages.
policies defined in [RFC3758] and [RFC7496] are:
o Limiting the time a user message is dealt with by the sender.
o Limiting the number of retransmissions for each fragment of a user
message. If the number of retransmissions is limited to 0, one
gets a service similar to UDP.
o Abandoning messages of lower priority in case of a send buffer
shortage.
SCTP supports multi-homing. Each SCTP endpoint uses a list of IP- SCTP supports multi-homing. Each SCTP endpoint uses a list of IP-
addresses and a single port number. These addresses can be any addresses and a single port number. These addresses can be any
mixture of IPv4 and IPv6 addresses. These addresses are negotiated mixture of IPv4 and IPv6 addresses. These addresses are negotiated
during the handshake and the address re-configuration extension during the handshake and the address re-configuration extension
specified in [RFC5061] in combination with [RFC4895] can be used to specified in [RFC5061] in combination with [RFC4895] can be used to
change these addresses in an authenticated way during the livetime of change these addresses in an authenticated way during the lifetime of
an SCTP association. This allows for transport layer mobility. an SCTP association. This allows for transport layer mobility.
Multiple addresses are used for improved resilience. If a remote Multiple addresses are used for improved resilience. If a remote
address becomes unreachable, the traffic is switched over to a address becomes unreachable, the traffic is switched over to a
reachable one, if one exists. [I-D.ietf-tsvwg-sctp-failover] reachable one, if one exists.
specifies a quicker failover operation reducing the latency of the
failover.
For securing user messages, the use of TLS over SCTP has been For securing user messages, the use of TLS over SCTP has been
specified in [RFC3436]. However, this solution does not support all specified in [RFC3436]. However, this solution does not support all
services provided by SCTP, such as un-ordered delivery or partial services provided by SCTP, such as un-ordered delivery or partial
reliability. Therefore, the use of DTLS over SCTP has been specified reliability. Therefore, the use of DTLS over SCTP has been specified
in [RFC6083] to overcome these limitations. When using DTLS over in [RFC6083] to overcome these limitations. When using DTLS over
SCTP, the application can use almost all services provided by SCTP. SCTP, the application can use almost all services provided by SCTP.
[I-D.ietf-tsvwg-natsupp] defines methods for endpoints and [I-D.ietf-tsvwg-natsupp] defines methods for endpoints and
middleboxes to provide support NAT for SCTP over IPv4. For legacy middleboxes to provide NAT traversal for SCTP over IPv4. For legacy
NAT traversal, [RFC6951] defines the UDP encapsulation of SCTP- NAT traversal, [RFC6951] defines the UDP encapsulation of SCTP-
packets. Alternatively, SCTP packets can be encapsulated in DTLS packets. Alternatively, SCTP packets can be encapsulated in DTLS
packets as specified in [I-D.ietf-tsvwg-sctp-dtls-encaps]. The packets as specified in [I-D.ietf-tsvwg-sctp-dtls-encaps]. The
latter encapsulation is used within the WebRTC context. latter encapsulation is used within the WebRTC context.
SCTP has a well-defined API, described in the next subsection. SCTP has a well-defined API, described in the next subsection.
4.3.2. Interface Description 3.3.2. Interface Description
[RFC4960] defines an abstract API for the base protocol. This API [RFC4960] defines an abstract API for the base protocol. This API
describes the following functions callable by the upper layer of describes the following functions callable by the upper layer of
SCTP: Initialize, Associate, Send, Receive, Receive Unsent Message, SCTP: Initialize, Associate, Send, Receive, Receive Unsent Message,
Receive Unacknowledged Message, Shutdown, Abort, SetPrimary, Status, Receive Unacknowledged Message, Shutdown, Abort, SetPrimary, Status,
Change Heartbeat, Request Heartbeat, Get SRTT Report, Set Failure Change Heartbeat, Request Heartbeat, Get SRTT Report, Set Failure
Threshold, Set Protocol Parameters, and Destroy. The following Threshold, Set Protocol Parameters, and Destroy. The following
notifications are provided by the SCTP stack to the upper layer: notifications are provided by the SCTP stack to the upper layer:
COMMUNICATION UP, DATA ARRIVE, SHUTDOWN COMPLETE, COMMUNICATION LOST, COMMUNICATION UP, DATA ARRIVE, SHUTDOWN COMPLETE, COMMUNICATION LOST,
COMMUNICATION ERROR, RESTART, SEND FAILURE, NETWORK STATUS CHANGE. COMMUNICATION ERROR, RESTART, SEND FAILURE, NETWORK STATUS CHANGE.
An extension to the BSD Sockets API is defined in [RFC6458] and An extension to the BSD Sockets API is defined in [RFC6458] and
covers: covers:
o the base protocol defined in [RFC4960]. The API allows control o the base protocol defined in [RFC4960]. The API allows control
over local addresses and port numbers and the primary path. over local addresses and port numbers and the primary path.
Furthermore the application has fine control about parameters like Furthermore the application has fine control about parameters like
retransmission thresholds, the path supervision parameters, the retransmission thresholds, the path supervision parameters, the
delayed acknowledgment timeout, and the fragmentation point. The delayed acknowledgment timeout, and the fragmentation point. The
API provides a mechanism to allow the SCTP stack to notify the API provides a mechanism to allow the SCTP stack to notify the
application about event if the application has requested them. application about events if the application has requested them.
These notifications provide Information about status changes of These notifications provide information about status changes of
the association and each of the peer addresses. In case of send the association and each of the peer addresses. In case of send
failures, including drop of messages sent unreliably, the failures, including drop of messages sent unreliably, the
application can also be notified and user messages can be returned application can also be notified and user messages can be returned
to the application. When sending user messages, the stream id, a to the application. When sending user messages, the stream id, a
payload protocol identifier, an indication whether ordered payload protocol identifier, an indication whether ordered
delivery is requested or not. These parameters can also be delivery is requested or not. These parameters can also be
provided on message reception. Additionally a context can be provided on message reception. Additionally a context can be
provided when sending, which can be use in case of send failures. provided when sending, which can be use in case of send failures.
The sending of arbitrary large user messages is supported. The sending of arbitrary large user messages is supported.
o the SCTP Partial Reliability extension defined in [RFC3758] to o the SCTP Partial Reliability extension defined in [RFC3758] to
specify for a user message the PR-SCTP policy and the policy specify for a user message the PR-SCTP policy and the policy
specific parameter. specific parameter. Examples of these policies defined in
[RFC3758] and [RFC7496] are:
* Limiting the time a user message is dealt with by the sender.
* Limiting the number of retransmissions for each fragment of a
user message. If the number of retransmissions is limited to
0, one gets a service similar to UDP.
* Abandoning messages of lower priority in case of a send buffer
shortage.
o the SCTP Authentication extension defined in [RFC4895] allowing to o the SCTP Authentication extension defined in [RFC4895] allowing to
manage the shared keys, the HMAC to use, set the chunk types which manage the shared keys, the HMAC to use, set the chunk types which
are only accepted in an authenticated way, and get the list of are only accepted in an authenticated way, and get the list of
chunks which are accepted by the local and remote end point in an chunks which are accepted by the local and remote end point in an
authenticated way. authenticated way.
o the SCTP Dynamic Address Reconfiguration extension defined in o the SCTP Dynamic Address Reconfiguration extension defined in
[RFC5061]. It allows to manually add and delete local addresses [RFC5061]. It allows to manually add and delete local addresses
for SCTP associations and the enabling of automatic address for SCTP associations and the enabling of automatic address
skipping to change at page 15, line 31 skipping to change at page 15, line 21
are specifically marked as notifications. are specifically marked as notifications.
New functions have been introduced to support the use of multiple New functions have been introduced to support the use of multiple
local and remote addresses. Additional SCTP-specific send and local and remote addresses. Additional SCTP-specific send and
receive calls have been defined to permit SCTP-specific information receive calls have been defined to permit SCTP-specific information
to be sent without using ancillary data in the form of additional to be sent without using ancillary data in the form of additional
cmsgs. These functions provide support for detecting partial cmsgs. These functions provide support for detecting partial
delivery of user messages and notifications. delivery of user messages and notifications.
The SCTP socket API allows a fine-grained control of the protocol The SCTP socket API allows a fine-grained control of the protocol
behaviour through an extensive set of socket options. behavior through an extensive set of socket options.
The SCTP kernel implementations of FreeBSD, Linux and Solaris follow The SCTP kernel implementations of FreeBSD, Linux and Solaris follow
mostly the specified extension to the BSD Sockets API for the base mostly the specified extension to the BSD Sockets API for the base
protocol and the corresponding supported protocol extensions. protocol and the corresponding supported protocol extensions.
4.3.3. Transport Features 3.3.3. Transport Features
The transport features provided by SCTP are: The transport features provided by SCTP are:
o unicast. o connection-oriented transport with feature negotiation and
application-to-port mapping,
o connection setup with feature negotiation and application-to-port
mapping.
o port multiplexing.
o Uni-or bidirectional communication.
o message-oriented delivery supporting multiple concurrent streams. o unicast transport,
o fully reliable, partially reliable, or unreliable delivery. o port multiplexing,
o ordered and unordered delivery within a stream. o uni- or bidirectional communication,
o user message fragmentation and reassembly. o message-oriented delivery with durable message framing supporting
multiple concurrent streams,
o support for stream scheduling prioritization. o fully reliable, partially reliable, or unreliable delivery (based
on user specified policy to handle abandoned user messages) with
drop notification,
o user message bundling. o ordered and unordered delivery within a stream,
o flow control using a window-based mechanism. o support for stream scheduling prioritization,
o congestion control using methods similar to TCP. o segmentation,
o user message bundling,
o strong error/misdelivery detection (CRC32c). o flow control using a window-based mechanism,
o transport layer multihoming for resilience. o congestion control using methods similar to TCP,
o transport layer mobility. o strong error detection (CRC32c),
o resilience to network failure and/or handover. o transport layer multihoming for resilience and mobility.
4.4. User Datagram Protocol (UDP) 3.4. User Datagram Protocol (UDP)
The User Datagram Protocol (UDP) [RFC0768] [RFC2460] is an IETF The User Datagram Protocol (UDP) [RFC0768] [RFC2460] is an IETF
standards track transport protocol. It provides a unidirectional standards track transport protocol. It provides a unidirectional
datagram protocol that preserves message boundaries. It provides no datagram protocol that preserves message boundaries. It provides no
error correction,congestion control, or flow control. It can be used error correction, congestion control, or flow control. It can be
to send broadcast datagrams (IPv4) or multicast datagrams (IPv4 and used to send broadcast datagrams (IPv4) or multicast datagrams (IPv4
IPv6), in addition to unicast and anycast datagrams. IETF guidance and IPv6), in addition to unicast and anycast datagrams. IETF
on the use of UDP is provided in {{I-D.ietf-tsvwg- rfc5405bis}}. UDP guidance on the use of UDP is provided in
is widely implemented and widely used by common applications, [I-D.ietf-tsvwg-rfc5405bis]. UDP is widely implemented and widely
including DNS. used by common applications, including DNS.
4.4.1. Protocol Description 3.4.1. Protocol Description
UDP is a connection-less protocol that maintains message boundaries, UDP is a connection-less protocol that maintains message boundaries,
with no connection setup or feature negotiation. The protocol uses with no connection setup or feature negotiation. The protocol uses
independent messages, ordinarily called datagrams. Each stream of independent messages, ordinarily called datagrams. It provides
messages is independently managed, therefore retransmission does not
hold back data sent using other logical streams. It provides
detection of payload errors and misdelivery of packets to an detection of payload errors and misdelivery of packets to an
unintended endpoint, either of which result in discard of received unintended endpoint, either of which result in discard of received
datagrams, with no indication to the user of the service. datagrams, with no indication to the user of the service.
It is possible to create IPv4 UDP datagrams with no checksum, and It is possible to create IPv4 UDP datagrams with no checksum, and
while this is generally discouraged [RFC1122] while this is generally discouraged [RFC1122]
[I-D.ietf-tsvwg-rfc5405bis], certain special cases permit this use. [I-D.ietf-tsvwg-rfc5405bis], certain special cases permit this use.
These datagrams rely on the IPv4 header checksum to protect from These datagrams rely on the IPv4 header checksum to protect from
misdelivery to an unintended endpoint. IPv6 does not by permit UDP misdelivery to an unintended endpoint. IPv6 does not permit UDP
datagrams with no checksum, although in certain cases this rule may datagrams with no checksum, although in certain cases this rule may
be relaxed [RFC6935]. The checksum support considerations for be relaxed [RFC6935].
omitting the checksum are defined in [RFC6936].
UDP does not provide reliability and does not provide retransmission. UDP does not provide reliability and does not provide retransmission.
This implies messages may be re-ordered, lost, or duplicated in This implies messages may be re-ordered, lost, or duplicated in
transit. Note that due to the relatively weak form of checksum used transit. Note that due to the relatively weak form of checksum used
by UDP, applications that require end to end integrity of data are by UDP, applications that require end to end integrity of data are
recommended to include a stronger integrity check of their payload recommended to include a stronger integrity check of their payload
data. data.
Because UDP provides no flow control, a receiving application that is Because UDP provides no flow control, a receiving application that is
unable to run sufficiently fast, or frequently, may miss messages. unable to run sufficiently fast, or frequently, may miss messages.
skipping to change at page 17, line 17 skipping to change at page 17, line 4
UDP does not provide reliability and does not provide retransmission. UDP does not provide reliability and does not provide retransmission.
This implies messages may be re-ordered, lost, or duplicated in This implies messages may be re-ordered, lost, or duplicated in
transit. Note that due to the relatively weak form of checksum used transit. Note that due to the relatively weak form of checksum used
by UDP, applications that require end to end integrity of data are by UDP, applications that require end to end integrity of data are
recommended to include a stronger integrity check of their payload recommended to include a stronger integrity check of their payload
data. data.
Because UDP provides no flow control, a receiving application that is Because UDP provides no flow control, a receiving application that is
unable to run sufficiently fast, or frequently, may miss messages. unable to run sufficiently fast, or frequently, may miss messages.
The lack of congestion handling implies UDP traffic may experience The lack of congestion handling implies UDP traffic may experience
loss when using an overloaded path, and may cause the loss of loss when using an overloaded path, and may cause the loss of
messages from other protocols (e.g., TCP) when sharing the same messages from other protocols (e.g., TCP) when sharing the same
network path. network path.
On transmission, UDP encapsulates each datagram into an IP packet, On transmission, UDP encapsulates each datagram into a single IP
which may in turn be fragmented by IP. Fragments are reassembled packet or several IP packet fragments. This allows a datagram to be
before delivery to the UDP receiver. larger than the effective path MTU. Fragments are reassembled before
delivery to the UDP receiver, making this transparent to the user of
the transport service. When the jumbograms are supported, larger
messages may be sent without performing fragmentation.
Applications that need to provide fragmentation or that have other Applications that need to provide fragmentation or that have other
requirements such as receiver flow control, congestion control, requirements such as receiver flow control, congestion control,
PathMTU discovery/PLPMTUD, support for ECN, etc need these to be PathMTU discovery/PLPMTUD, support for ECN, etc. need these to be
provided by protocols operating over UDP [I-D.ietf-tsvwg-rfc5405bis]. provided by protocols operating over UDP [I-D.ietf-tsvwg-rfc5405bis].
4.4.2. Interface Description 3.4.2. Interface Description
[RFC0768] describes basic requirements for an API for UDP. Guidance [RFC0768] describes basic requirements for an API for UDP. Guidance
on use of common APIs is provided in [I-D.ietf-tsvwg-rfc5405bis]. on use of common APIs is provided in [I-D.ietf-tsvwg-rfc5405bis].
A UDP endpoint consists of a tuple of (IP address, port number). A UDP endpoint consists of a tuple of (IP address, port number). De-
Demultiplexing using multiple abstract endpoints (sockets) on the multiplexing using multiple abstract endpoints (sockets) on the same
same IP address are supported. The same socket may be used by a IP address is supported. The same socket may be used by a single
single server to interact with multiple clients (note: this behavior server to interact with multiple clients (note: this behavior differs
differs from TCP, which uses a pair of tuples to identify a from TCP, which uses a pair of tuples to identify a connection).
connection). Multiple server instances (processes) that bind the Multiple server instances (processes) that bind to the same socket
same socket can cooperate to service multiple clients- the socket can cooperate to service multiple clients. The socket implementation
implementation arranges to not duplicate the same received unicast arranges to not duplicate the same received unicast message to
message to multiple server processes. multiple server processes.
Many operating systems also allow a UDP socket to be "connected", Many operating systems also allow a UDP socket to be "connected",
i.e., to bind a UDP socket to a specific (remote) UDP endpoint. i.e., to bind a UDP socket to a specific (remote) UDP endpoint.
Unlike TCP's connect primitive, for UDP, this is only a local Unlike TCP's connect primitive, for UDP, this is only a local
operation that serves to simplify the local send/receive functions operation that serves to simplify the local send/receive functions
and to filter the traffic for the specified addresses and ports and to filter the traffic for the specified addresses and ports
[I-D.ietf-tsvwg-rfc5405bis]. [I-D.ietf-tsvwg-rfc5405bis].
4.4.3. Transport Features 3.4.3. Transport Features
The transport features provided by UDP are: The transport features provided by UDP are:
o unicast. o unicast, multicast, anycast, or IPv4 broadcast transport,
o multicast, anycast, or IPv4 broadcast.
o port multiplexing. A receiving port can be configured to receive
datagrams from multiple senders.
o message-oriented delivery. o port multiplexing (where a receiving port can be configured to
receive datagrams from multiple senders),
o Uni-or bidirectional communication. Transmission in each o message-oriented delivery,
direction is independent.
o non-reliable delivery. o uni- or bidirectional communication where the transmissions in
each direction are independent,
o non-ordered delivery. o non-reliable delivery,
o error detection: a segment checksum verifies delivery to the o unordered delivery,
correct endpoint and integrity of the data. This checksum is
optional for IPv4, and optional under specific conditions for IPv6
where all or none of the payload data is protected.
o IPv6 jumbograms. o error detection (implemented using a segment checksum to verify
delivery to the correct endpoint and integrity of the data;
optional for IPv4 and optional under specific conditions for IPv6
where all or none of the payload data is protected),
4.5. Lightweight User Datagram Protocol (UDP-Lite) 3.5. Lightweight User Datagram Protocol (UDP-Lite)
The Lightweight User Datagram Protocol (UDP-Lite) [RFC3828] is an The Lightweight User Datagram Protocol (UDP-Lite) [RFC3828] is an
IETF standards track transport protocol. It provides a IETF standards track transport protocol. It provides a
unidirectional, datagram protocol that preserves message boundaries. unidirectional, datagram protocol that preserves message boundaries.
IETF guidance on the use of UDP- Lite is provided in IETF guidance on the use of UDP- Lite is provided in
[I-D.ietf-tsvwg-rfc5405bis]. [I-D.ietf-tsvwg-rfc5405bis]. A UDP-Lite service may support IPv4
broadcast, multicast, anycast and unicast, and IPv6 multicast,
4.5.1. Protocol Description anycast and unicast.
Like UDP, UDP-Lite is a connection-less datagram protocol, with no
connection setup or feature negotiation. It changes the semantics of
the UDP "payload length" field to that of a "checksum coverage
length" field, and is identified by a different IP protocol/next-
header value. Otherwise, UDP-Lite is semantically identical to UDP.
Applications using UDP-Lite therefore cannot make assumptions
regarding the correctness of the data received in the insensitive
part of the UDP-Lite payload.
In the same way as for UDP, mechanisms for receiver flow control,
congestion control, PMTU or PLPMTU discovery, support for ECN, etc
need to be provided by upper layer protocols
[I-D.ietf-tsvwg-rfc5405bis].
Examples of use include a class of applications that can derive Examples of use include a class of applications that can derive
benefit from having partially-damaged payloads delivered, rather than benefit from having partially-damaged payloads delivered, rather than
discarded. One use is to support error tolerate payload corruption discarded. One use is to support error tolerate payload corruption
when used over paths that include error-prone links, another when used over paths that include error-prone links, another
application is when header integrity checks are required, but payload application is when header integrity checks are required, but payload
integrity is provided by some other mechanism (e.g., [RFC6936]). integrity is provided by some other mechanism (e.g., [RFC6936]).
A UDP-Lite service may support IPv4 broadcast, multicast, anycast and 3.5.1. Protocol Description
unicast, and IPv6 multicast, anycast and unicast.
4.5.2. Interface Description Like UDP, UDP-Lite is a connection-less datagram protocol, with no
connection setup or feature negotiation. It changes the semantics of
the UDP "payload length" field to that of a "checksum coverage
length" field, and is identified by a different IP protocol/next-
header value. The "checksum coverage length" field specifies the
intended checksum coverage, with the remaining unprotected part of
the payload called the "error-insensitive part". Applications using
UDP-Lite therefore cannot make assumptions regarding the correctness
of the data received in the insensitive part of the UDP-Lite payload.
Otherwise, UDP-Lite is semantically identical to UDP. In the same
way as for UDP, mechanisms for receiver flow control, congestion
control, PMTU or PLPMTU discovery, support for ECN, etc. needs to be
provided by upper layer protocols [I-D.ietf-tsvwg-rfc5405bis].
3.5.2. Interface Description
There is no API currently specified in the RFC Series, but guidance There is no API currently specified in the RFC Series, but guidance
on use of common APIs is provided in [I-D.ietf-tsvwg-rfc5405bis]. on use of common APIs is provided in [I-D.ietf-tsvwg-rfc5405bis].
The interface of UDP-Lite differs from that of UDP by the addition of The interface of UDP-Lite differs from that of UDP by the addition of
a single (socket) option that communicates a checksum coverage length a single (socket) option that communicates a checksum coverage length
value: at the sender, this specifies the intended checksum coverage, value. The checksum coverage may also be made visible to the
with the remaining unprotected part of the payload called the "error- application via the UDP-Lite MIB module [RFC5097].
insensitive part". The checksum coverage may also be made visible to
the application via the UDP-Lite MIB module [RFC5097].
4.5.3. Transport Features 3.5.3. Transport Features
The transport features provided by UDP-Lite are: The transport features provided by UDP-Lite are:
o unicast. o unicast, multicast, anycast, or IPv4 broadcast transport (as for
UDP),
o multicast, anycast, or IPv4 broadcast.
o port multiplexing (as for UDP).
o message-oriented delivery (as for UDP). o port multiplexing (as for UDP),
o Uni-or bidirectional communication. Transmission in each o message-oriented delivery (as for UDP),
direction is independent.
o non-reliable delivery (as for UDP). o Uni- or bidirectional communication where the transmissions in
each direction are independent (as for UDP),
o non-ordered delivery (as for UDP). o non-reliable delivery (as for UDP),
o misdelivery detection (the checksum always provides protection o non-ordered delivery (as for UDP),
from misdelivery).
o partial or full integrity protection. The checksum coverage field o partial or full payload error detection (where the checksum
indicates the size of the payload data covered by the checksum. coverage field indicates the size of the payload data covered by
the checksum).
4.6. Datagram Congestion Control Protocol (DCCP) 3.6. Datagram Congestion Control Protocol (DCCP)
Datagram Congestion Control Protocol (DCCP) [RFC4340] is an IETF Datagram Congestion Control Protocol (DCCP) [RFC4340] is an IETF
standards track bidirectional transport protocol that provides standards track bidirectional transport protocol that provides
unicast connections of congestion-controlled messages without unicast connections of congestion-controlled messages without
providing reliability. providing reliability.
The DCCP Problem Statement describes the goals that DCCP sought to The DCCP Problem Statement describes the goals that DCCP sought to
address [RFC4336]. It is suitable for applications that transfer address [RFC4336]: It is suitable for applications that transfer
fairly large amounts of data and that can benefit from control over fairly large amounts of data and that can benefit from control over
the trade off between timeliness and reliability [RFC4336]. the trade off between timeliness and reliability [RFC4336].
DCCP offers low overhead, and many characteristics common to UDP, but DCCP offers low overhead, and many characteristics common to UDP, but
can avoid "re-inventing the wheel" each time a new multimedia can avoid "re-inventing the wheel" each time a new multimedia
application emerges. Specifically it includes core functions application emerges. Specifically it includes core transport
(feature negotiation, path state management, RTT calculation, PMTUD, functions (feature negotiation, path state management, RTT
etc): This allows applications to use a compatible method defining calculation, PMTUD, etc.): DCCP applications select how they send
how they send packets and where suitable to choose common algorithms packets and, where suitable, choose common algorithms to manage their
to manage their functions. Examples of suitable applications include functions. Examples of applications that can benefit from such
interactive applications, streaming media or on-line games [RFC4336]. transport services include interactive applications, streaming media,
or on-line games [RFC4336].
4.6.1. Protocol Description 3.6.1. Protocol Description
DCCP is a connection-oriented datagram protocol, providing a three- DCCP is a connection-oriented datagram protocol, providing a three-
way handshake to allow a client and server to set up a connection, way handshake to allow a client and server to set up a connection,
and mechanisms for orderly completion and immediate teardown of a and mechanisms for orderly completion and immediate teardown of a
connection. The protocol is defined by a family of RFCs. connection.
A DCCP protocol instance can be extended [RFC4340] and tuned using
additional features. Some features are sender-side only, requiring
no negotiation with the receiver; some are receiver-side only; and
some are explicitly negotiated during connection setup.
DCCP uses a Connect packet to initiate a session, and permits each
endpoint to choose the features it wishes to support. Simultaneous
open [RFC5596], as in TCP, can enable interoperability in the
presence of middleboxes. The Connect packet includes a Service Code
[RFC5595] that identifies the application or protocol using DCCP,
providing middleboxes with information about the intended use of a
connection.
DCCP service is unicast-only.
It provides multiplexing to multiple sockets at each endpoint using It provides multiplexing to multiple sockets at each endpoint using
port numbers. An active DCCP session is identified by its four-tuple port numbers. An active DCCP session is identified by its four-tuple
of local and remote IP addresses and local port and remote port of local and remote IP addresses and local port and remote port
numbers. At connection setup, DCCP also exchanges the service code numbers.
[RFC5595], a mechanism that allows transport instantiations to
indicate the service treatment that is expected from the network.
The protocol segments data into messages, typically sized to fit in The protocol segments data into messages, typically sized to fit in
IP packets, but which may be fragmented providing they are less than IP packets, but which may be fragmented providing they are smaller
the maximum packet size. A DCCP interface allows applications to than the maximum packet size. A DCCP interface allows applications
request fragmentation for packets larger than PMTU, but not larger to request fragmentation for packets larger than PMTU, but not larger
than the maximum packet size allowed by the current congestion than the maximum packet size allowed by the current congestion
control mechanism (CCMPS) [RFC4340]. control mechanism (CCMPS) [RFC4340].
Each message is identified by a sequence number. The sequence number Each message is identified by a sequence number. The sequence number
is used to identify segments in acknowledgments, to detect is used to identify segments in acknowledgments, to detect
unacknowledged segments, to measure RTT, etc. The protocol may unacknowledged segments, to measure RTT, etc. The protocol may
support ordered or unordered delivery of data, and does not itself support unordered delivery of data, and does not itself provide
provide retransmission. DCCP supports reduced checksum coverage, a retransmission. DCCP supports reduced checksum coverage, a partial
partial integrity mechanism similar to UDP-Lite. There is also a payload protection mechanism similar to UDP-Lite. There is also a
Data Checksum option that when enabled, contains a strong CRC, to Data Checksum option, which when enabled, contains a strong CRC, to
enable endpoints to detect application data corruption - similar to enable endpoints to detect application data corruption.
SCTP.
Receiver flow control is supported, which limits the amount of Receiver flow control is supported, which limits the amount of
unacknowledged data that can be outstanding at a given time. unacknowledged data that can be outstanding at a given time.
A DCCP protocol instance can be extended [RFC4340] and tuned using DCCP supports negotiation of the congestion control profile between
additional features. Some features are sender-side only, requiring endpoints, to provide plug-and-play congestion control mechanisms.
no negotiation with the receiver; some are receiver-side only; and Examples of specified profiles include "TCP-like" [RFC4341], "TCP-
some are explicitly negotiated during connection setup. friendly" [RFC4342], and "TCP-friendly for small packets" [RFC5622].
Additional mechanisms are recorded in an IANA registry.
DCCP service is unicast-only.
It supports negotiation of the congestion control profile, to provide
plug- and-play congestion control mechanisms. Examples of specified
profiles include "TCP-like" [RFC4341], "TCP-friendly" [RFC4342], and
"TCP-friendly for small packets" [RFC5622]. Additional mechanisms
are recorded in an IANA registry.
DCCP uses a Connect packet to initiate a session, and permits half-
connections that allow each client to choose the features it wishes
to support. Simultaneous open [RFC5596], as in TCP, can enable
interoperability in the presence of middleboxes. The Connect packet
includes a Service Code field [RFC5595] designed to allow middleboxes
and endpoints to identify the characteristics required by a session.
A lightweight UDP-based encapsulation (DCCP-UDP) has been defined A lightweight UDP-based encapsulation (DCCP-UDP) has been defined
[RFC6773] that permits DCCP to be used over paths where DCCP is not [RFC6773] that permits DCCP to be used over paths where DCCP is not
natively supported. Support in NAPT/NATs is defined in [RFC4340] and natively supported. Support for DCCP in NAPT/NATs is defined in
[RFC5595]. [RFC4340] and [RFC5595]. Upper layer protocols specified on top of
DCCP include DTLS [RFC5595], RTP [RFC5672], ICE/SDP [RFC6773].
Upper layer protocols specified on top of DCCP include DTLS
[RFC5595], RTP [RFC5672], ICE/SDP [RFC6773].
A common packet format has allowed tools to evolve that can read and
interpret DCCP packets (e.g., Wireshark).
4.6.2. Interface Description 3.6.2. Interface Description
API characteristics include: - Datagram transmission. - Notification Functions expected for a DCCP API include: Open, Close and Management
of the current maximum packet size. - Send and reception of zero- of the progress a DCCP connection. The Open function provides
length payloads. - Slow Receiver flow control at a receiver. - feature negotiation, selection of an appropriate CCID for congestion
ability to detect a slow receiver at the sender. control and other parameters associated with the DCCP connection. A
function allows an application to send DCCP datagrams, including
setting the required checksum coverage, and any required options.
(DCCP permits sending datagrams with a zero-length payload.) A
function allows reception of data, including indicating if the data
was used or dropped. Functions can also make the status of a
connection visible to an application, including detection of the
maximum packet size and the ability to perform flow control by
detecting a slow receiver at the sender.
There is no API currently specified in the RFC Series. There is no API currently specified in the RFC Series.
4.6.3. Transport Features 3.6.3. Transport Features
The transport features provided by DCCP are: The transport features provided by DCCP are:
o unicast transport. o unicast transport,
o connection setup with feature negotiation and application-to-port
mapping.
o Service Codes. Identifies the upper layer service to the endpoint
and network.
o port multiplexing.
o Uni-or bidirectional communication. o connection-oriented communication with feature negotiation and
application-to-port mapping,
o message-oriented delivery. o signaling of application class for middlebox support (implemented
using Service Codes),
o non-reliable delivery. o port multiplexing,
o ordered delivery. o uni-or bidirectional communication,
o message-oriented delivery,
o flow control. The slow receiver function allows a receiver to o unreliable delivery with drop notification,
control the rate of the sender.
o drop notification. Allows a receiver to notify which datagrams o unordered delivery,
were not delivered to the peer upper layer protocol.
o timestamps. o flow control (implemented using the slow receiver function)
o partial and full integrity protection (with optional strong o partial and full payload error detection (with optional strong
integrity check). integrity check).
4.7. Internet Control Message Protocol (ICMP) 3.7. Internet Control Message Protocol (ICMP)
The Internet Control Message Protocol (ICMP) [RFC0792] for IPv4 and The Internet Control Message Protocol (ICMP) [RFC0792] for IPv4 and
[RFC4433] for IPv6 are IETF standards track protocols. ICMP for IPv6 [RFC4433] are IETF standards track protocols. It is a
connection-less unidirectional protocol that delivers individual
ICMP is a connection-less unidirectional protocol that delivers messages, without error correction, congestion control, or flow
individual messages, without error correction, congestion control, or control. Messages may be sent as unicast, IPv4 broadcast or
flow control. Messages may be sent as unicast, IPv4 broadcast or
multicast datagrams (IPv4 and IPv6), in addition to anycast multicast datagrams (IPv4 and IPv6), in addition to anycast
datagrams. datagrams.
4.7.1. Protocol Description Transport Protocols and upper layer protocols can use received ICMP
messages to help them take appropriate decisions when network or
endpoint errors are reported. For example, to implement, ICMP-based
Path MTU discovery [RFC1191][RFC1981] or assist in Packetization
Layer Path MTU Discovery (PMTUD) [RFC4821]. Such reactions to
received messages need to protect from off-path data injection
[I-D.ietf-tsvwg-rfc5405bis], to avoid an application receiving
packets created by an unauthorized third party. An application
therefore needs to ensure that all messages are appropriately
validated, by checking the payload of the messages to ensure these
are received in response to actually transmitted traffic (e.g., a
reported error condition that corresponds to a UDP datagram or TCP
segment was actually sent by the application). This requires context
[RFC6056], such as local state about communication instances to each
destination (e.g., in the TCP, DCCP, or SCTP protocols). This state
is not always maintained by UDP-based applications
[I-D.ietf-tsvwg-rfc5405bis].
ICMP is a connection-less unidirectional protocol that delivers 3.7.1. Protocol Description
individual messages. The protocol uses independent messages,
ordinarily called datagrams. Each message is required to carry a ICMP is a connection-less unidirectional protocol, It delivers
checksum as an integrity check and to protect from misdelivery to an independent messages, called datagrams. Each message is required to
unintended endpoint. carry a checksum as an integrity check and to protect from mis-
delivery to an unintended endpoint.
ICMP messages typically relay diagnostic information from an endpoint ICMP messages typically relay diagnostic information from an endpoint
[RFC1122] or network device [RFC1716] addressed to the sender of a [RFC1122] or network device [RFC1716] addressed to the sender of a
flow. This usually contains the network protocol header of a packet flow. This usually contains the network protocol header of a packet
that encountered a reported issue. Some formats of messages can also that encountered a reported issue. Some formats of messages can also
carry other payload data. Each message carries an integrity check carry other payload data. Each message carries an integrity check
calculated in the same way as for UDP, this checksum is not optional. calculated in the same way as for UDP, this checksum is not optional.
The RFC series defines additional IPv6 message formats to support a The RFC series defines additional IPv6 message formats to support a
range of uses. In the case of IPv6 the protocol incorporates range of uses. In the case of IPv6 the protocol incorporates
neighbor discovery [RFC2461] [RFC3971]} (provided by ARP for IPv4) neighbor discovery [RFC2461] [RFC3971]} (provided by ARP for IPv4)
and the Multicast Listener Discovery (MLD) [RFC2710] group management and the Multicast Listener Discovery (MLD) [RFC2710] group management
functions (provided by IGMP for IPv4). functions (provided by IGMP for IPv4).
Reliable transmission can not be assumed. A receiving application Reliable transmission can not be assumed. A receiving application
that is unable to run sufficiently fast, or frequently, may miss that is unable to run sufficiently fast, or frequently, may miss
messages since there is no flow or congestion control. In addition messages since there is no flow or congestion control. In addition
some network devices rate-limit ICMP messages. some network devices rate-limit ICMP messages.
Transport Protocols and upper layer protocols can use received ICMP 3.7.2. Interface Description
messages to help them take appropriate decisions when network or
endpoint errors are reported. For example to implement, ICMP-based ICMP processing is integrated in many connection-oriented transports,
Path MTU discovery [RFC1191][RFC1981] or assist in Packetization but like other functions needs to be provided by an upper-layer
Layer Path MTU Discovery (PMTUD) [RFC4821]. Such reactions to protocol when using UDP and UDP-Lite.
received messages need to protects from off-path data injection
[I-D.ietf-tsvwg-rfc5405bis], avoiding an application receiving On some stacks, a bound socket also allows a UDP application to be
packets that were created by an unauthorized third party. An notified when ICMP error messages are received for its transmissions
application therefore needs to ensure that all messages are
appropriately validated, by checking the payload of the messages to
ensure these are received in response to actually transmitted traffic
(e.g., a reported error condition that corresponds to a UDP datagram
or TCP segment was actually sent by the application). This requires
context [RFC6056], such as local state about communication instances
to each destination (e.g., in the TCP, DCCP, or SCTP protocols).
This state is not always maintained by UDP-based applications
[I-D.ietf-tsvwg-rfc5405bis]. [I-D.ietf-tsvwg-rfc5405bis].
Any response to ICMP error messages ought to be robust to temporary Any response to ICMP error messages ought to be robust to temporary
routing failures (sometimes called "soft errors"), e.g., transient routing failures (sometimes called "soft errors"), e.g., transient
ICMP "unreachable" messages ought to not normally cause a ICMP "unreachable" messages ought to not normally cause a
communication abort [RFC5461] [I-D.ietf-tsvwg-rfc5405bis]. communication abort [RFC5461] [I-D.ietf-tsvwg-rfc5405bis].
4.7.2. Interface Description 3.7.3. Transport Features
ICMP processing is integrated into many connection-oriented
transports, but like other functions needs to be provided by an
upper-layer protocol when using UDP and UDP-Lite. On some stacks, a
bound socket also allows a UDP application to be notified when ICMP
error messages are received for its transmissions
[I-D.ietf-tsvwg-rfc5405bis].
4.7.3. Transport Features
The transport features provided by ICMP are:
o unidirectional.
o multicast, anycast and IP4 broadcast.
o message-oriented delivery.
o non-reliable delivery.
o non-ordered delivery.
o error and misdelivery detection (checksum). ICMP does not provide any transport service directly to applications.
Used together with other transport protocols, it provides
transmission of control, error, and measurement data between
endpoints, or from devices along the path to one endpoint.
4.8. Realtime Transport Protocol (RTP) 3.8. Realtime Transport Protocol (RTP)
RTP provides an end-to-end network transport service, suitable for RTP provides an end-to-end network transport service, suitable for
applications transmitting real-time data, such as audio, video or applications transmitting real-time data, such as audio, video or
data, over multicast or unicast network services, including TCP, UDP, data, over multicast or unicast transport services, including TCP,
UDP-Lite, or DCCP. UDP, UDP-Lite, DCCP, TLS and DTLS.
4.8.1. Protocol Description 3.8.1. Protocol Description
The RTP standard [RFC3550] defines a pair of protocols, RTP and the The RTP standard [RFC3550] defines a pair of protocols, RTP and the
Real Time Control Protocol, RTCP. The transport does not provide RTP control protocol, RTCP. The transport does not provide
connection setup, instead relying on out-of-band techniques or connection setup, instead relying on out-of-band techniques or
associated control protocols to setup, negotiate parameters or tear associated control protocols to setup, negotiate parameters or tear
down a session. down a session.
An RTP sender encapsulates audio/video data into RTP packets to An RTP sender encapsulates audio/video data into RTP packets to
transport media streams. The RFC-series specifies RTP media formats transport media streams. The RFC-series specifies RTP payload
allow packets to carry a wide range of media, and specifies a wide formats that allow packets to carry a wide range of media, and
range of multiplexing, error control and other support mechanisms. specifies a wide range of multiplexing, error control and other
support mechanisms.
If a frame of media data is large, it will be fragmented into several If a frame of media data is large, it will be fragmented into several
RTP packets. Likewise, several small frames may be bundled into a RTP packets. Likewise, several small frames may be bundled into a
single RTP packet. RTP may run over a congestion-controlled or non- single RTP packet.
congestion-controlled transport protocol.
An RTP receiver collects RTP packets from network, validates them for An RTP receiver collects RTP packets from the network, validates them
correctness, and sends them to the media decoder input-queue. for correctness, and sends them to the media decoder input-queue.
Missing packet detection is performed by the channel decoder. The Missing packet detection is performed by the channel decoder. The
play-out buffer is ordered by time stamp and is used to reorder play-out buffer is ordered by time stamp and is used to reorder
packets. Damaged frames may be repaired before the media payloads packets. Damaged frames may be repaired before the media payloads
are decompressed to display or store the data. are decompressed to display or store the data. Some uses of RTP are
able to exploit the partial payload protection features offered by
DCCP and UDP-Lite.
RTCP is a control protocol that works alongside a RTP flow. Both the RTCP is a control protocol that works alongside an RTP flow. Both
RTP sender and receiver can send RTCP report packets. This is used the RTP sender and receiver will send RTCP report packets. This is
to periodically send control information and report performance. used to periodically send control information and report performance.
Based on received RTCP feedback, an RTP sender can adjust the Based on received RTCP feedback, an RTP sender can adjust the
transmission, e.g., perform rate adaptation at the application layer transmission, e.g., perform rate adaptation at the application layer
in the case of congestion. in the case of congestion.
An RTCP receiver report (RTCP RR) is returned to the sender An RTCP receiver report (RTCP RR) is returned to the sender
periodically to report key parameters (e.g, the fraction of packets periodically to report key parameters (e.g, the fraction of packets
lost in the last reporting interval, the cumulative number of packets lost in the last reporting interval, the cumulative number of packets
lost, the highest sequence number received, and the inter-arrival lost, the highest sequence number received, and the inter-arrival
jitter). The RTCP RR packets also contain timing information that jitter). The RTCP RR packets also contain timing information that
allows the sender to estimate the network round trip time (RTT) to allows the sender to estimate the network round trip time (RTT) to
the receivers. the receivers.
The interval between reports sent from each receiver tends to be on The interval between reports sent from each receiver tends to be on
the order of a few seconds on average, although this varies with the the order of a few seconds on average, although this varies with the
session rate, and sub-second reporting intervals are possible for session rate, and sub-second reporting intervals are possible for
high rate sessions. The interval is randomized to avoid high rate sessions. The interval is randomized to avoid
synchronization of reports from multiple receivers. synchronization of reports from multiple receivers.
4.8.2. Interface Description 3.8.2. Interface Description
There is no standard application programming interface defined for There is no standard application programming interface defined for
RTP or RTCP. Implementations are typically tightly integrated with a RTP or RTCP. Implementations are typically tightly integrated with a
particular application, and closely follow the principles of particular application, and closely follow the principles of
application level framing and integrated layer processing [ClarkArch] application level framing and integrated layer processing [ClarkArch]
in media processing [RFC2736], error recovery and concealment, rate in media processing [RFC2736], error recovery and concealment, rate
adaptation, and security [RFC7202]. Accordingly, RTP implementations adaptation, and security [RFC7202]. Accordingly, RTP implementations
tend to be targeted at particular application domains (e.g., voice- tend to be targeted at particular application domains (e.g., voice-
over-IP, IPTV, or video conferencing), with a feature set optimised over-IP, IPTV, or video conferencing), with a feature set optimized
for that domain, rather than being general purpose implementations of for that domain, rather than being general purpose implementations of
the protocol. the protocol.
4.8.3. Transport Features 3.8.3. Transport Features
The transport features provided by RTP are: The transport features provided by RTP are:
o unicast transport. o unicast, multicast or IPv4 broadcast (provided by lower layer
protocol),
o multicast, anycast or IPv4 broadcast.
o port multiplexing.
o Uni-or bidirectional communication.
o message-oriented delivery.
o associated protocols for connection setup with feature negotiation o port multiplexing (provided by lower layer protocol),
and application-to-port mapping.
o support for media types and other extensions. o uni- or bidirectional communication (provided by lower layer
protocol),
o a range of reliability functions, including the possibility of o message-oriented delivery with support for media types and other
using packet erasure coding. extensions,
o segmentation and aggregation. o reliable delivery when using erasure coding or unreliable delivery
with drop notification (if supported by lower layer protocol),
o performance reporting. o connection setup with feature negotiation (using associated
protocols) and application-to-port mapping (provided by lower
layer protocol),
o drop notification. o segmentation,
o timestamps. o performance metric reporting (using associated protocols).
4.9. File Delivery over Unidirectional Transport/Asynchronous Layered 3.9. File Delivery over Unidirectional Transport/Asynchronous Layered
Coding Reliable Multicast (FLUTE/ALC) Coding Reliable Multicast (FLUTE/ALC)
FLUTE/ALC is an IETF standards track protocol specified in [RFC6726] FLUTE/ALC is an IETF standards track protocol specified in [RFC6726]
and [RFC5775]. Asynchronous Layer Coding (ALC) provides an and [RFC5775]. It provides object-oriented delivery of discrete data
underlying reliable transport service and FLUTE a file-oriented or files. Asynchronous Layer Coding (ALC) provides an underlying
specialization of the ALC service (e.g., to carry associated reliable transport service and FLUTE a file-oriented specialization
metadata). The [RFC6726] and [RFC5775] protocols are non-backward- of the ALC service (e.g., to carry associated metadata). The
compatible updates of the [RFC3926] and [RFC3450] experimental [RFC6726] and [RFC5775] protocols are non-backward-compatible updates
protocols; these experimental protocols are currently largely of the [RFC3926] and [RFC3450] experimental protocols; these
deployed in the 3GPP Multimedia Broadcast and Multicast Services experimental protocols are currently largely deployed in the 3GPP
(MBMS) (see [MBMS], section 7) and similar contexts (e.g., the Multimedia Broadcast and Multicast Services (MBMS) (see [MBMS],
Japanese ISDB-Tmm standard). section 7) and similar contexts (e.g., the Japanese ISDB-Tmm
standard).
The FLUTE/ALC protocol has been designed to support massively The FLUTE/ALC protocol has been designed to support massively
scalable reliable bulk data dissemination to receiver groups of scalable reliable bulk data dissemination to receiver groups of
arbitrary size using IP Multicast over any type of delivery network, arbitrary size using IP Multicast over any type of delivery network,
including unidirectional networks (e.g., broadcast wireless including unidirectional networks (e.g., broadcast wireless
channels). However, the FLUTE/ALC protocol also supports point-to- channels). However, the FLUTE/ALC protocol also supports point-to-
point unicast transmissions. point unicast transmissions.
FLUTE/ALC bulk data dissemination has been designed for discrete file FLUTE/ALC bulk data dissemination has been designed for discrete file
or memory-based "objects". Transmissions happen either in push mode, or memory-based "objects". Although FLUTE/ALC is not well adapted to
where content is sent once, or in on-demand mode, where content is byte- and message-streaming, there is an exception: FLUTE/ALC is used
continuously sent during periods of time that can largely exceed the to carry 3GPP Dynamic Adaptive Streaming over HTTP (DASH) when
average time required to download the session objects (see [RFC5651], scalability is a requirement (see [MBMS], section 5.6).
section 4.2).
Although FLUTE/ALC is not well adapted to byte- and message- FLUTE/ALC's reliability, delivery mode, congestion control, and flow/
streaming, there is an exception: FLUTE/ALC is used to carry 3GPP rate control mechanisms can be separately controlled to meet
Dynamic Adaptive Streaming over HTTP (DASH) when scalability is a different application needs. Section 4.1 of
requirement (see [MBMS], section 5.6). In that case, each Audio/
Video segment is transmitted as a distinct FLUTE/ALC object in push
mode. FLUTE/ALC uses packet erasure coding (also known as
Application-Level Forward Erasure Correction, or AL-FEC) in a
proactive way. The goal of using AL-FEC is both to increase the
robustness in front of packet erasures and to improve the efficiency
of the on-demand service. FLUTE/ALC transmissions can be governed by
a congestion control mechanism such as the "Wave and Equation Based
Rate Control" (WEBRC) [RFC3738] when FLUTE/ALC is used in a layered
transmission manner, with several session channels over which ALC
packets are sent. However many FLUTE/ALC deployments target pre-
provisioned networks and involve only Constant Bit Rate (CBR)
channels with no competing flows, for which a sender-based rate
control mechanism is sufficient. In any case, FLUTE/ALC's
reliability, delivery mode, congestion control, and flow/rate control
mechanisms are distinct components that can be separately controlled
to meet different application needs. Section 4.1 of
[I-D.ietf-tsvwg-rfc5405bis] describes multicast congestion control [I-D.ietf-tsvwg-rfc5405bis] describes multicast congestion control
requirements for UDP. requirements for UDP.
4.9.1. Protocol Description 3.9.1. Protocol Description
The FLUTE/ALC protocol works on top of UDP (though it could work on The FLUTE/ALC protocol works on top of UDP (though it could work on
top of any datagram delivery transport protocol), without requiring top of any datagram delivery transport protocol), without requiring
any connectivity from receivers to the sender. Purely unidirectional any connectivity from receivers to the sender. Purely unidirectional
networks are therefore supported by FLUTE/ALC. This guarantees networks are therefore supported by FLUTE/ALC. This guarantees
scalability to an unlimited number of receivers in a session, since scalability to an unlimited number of receivers in a session, since
the sender behaves exactly the same regardless of the number of the sender behaves exactly the same regardless of the number of
receivers. receivers.
FLUTE/ALC supports the transfer of bulk objects such as file or in- FLUTE/ALC supports the transfer of bulk objects such as file or in-
memory content, using either a push or an on-demand mode. in push memory content, using either a push or an on-demand mode. in push
mode, content is sent once to the receivers, while in on-demand mode, mode, content is sent once to the receivers, while in on-demand mode,
content is sent continuously during periods of time that can greatly content is sent continuously during periods of time that can greatly
exceed the average time required to download the session objects. exceed the average time required to download the session objects (see
[RFC5651], section 4.2).
This enables receivers to join a session asynchronously, at their own This enables receivers to join a session asynchronously, at their own
discretion, receive the content and leave the session. In this case, discretion, receive the content and leave the session. In this case,
data content is typically sent continuously, in loops (also known as data content is typically sent continuously, in loops (also known as
"carousels"). FLUTE/ALC also supports the transfer of an object "carousels"). FLUTE/ALC also supports the transfer of an object
stream, with loose real-time constraints. This is particularly stream, with loose real-time constraints. This is particularly
useful to carry 3GPP DASH when scalability is a requirement and useful to carry 3GPP DASH when scalability is a requirement and
unicast transmissions over HTTP cannot be used ([MBMS], section 5.6). unicast transmissions over HTTP cannot be used ([MBMS], section 5.6).
In this case, packets are sent in sequence using push mode. FLUTE/ In this case, packets are sent in sequence using push mode. FLUTE/
ALC is not well adapted to byte- and message-streaming and other ALC is not well adapted to byte- and message-streaming and other
skipping to change at page 28, line 28 skipping to change at page 27, line 21
delivery service. Each object of the FLUTE/ALC session is described delivery service. Each object of the FLUTE/ALC session is described
in a dedicated entry of a File Delivery Table (FDT), using an XML in a dedicated entry of a File Delivery Table (FDT), using an XML
format (see [RFC6726], section 3.2). This metadata can include, but format (see [RFC6726], section 3.2). This metadata can include, but
is not restricted to, a URI attribute (to identify and locate the is not restricted to, a URI attribute (to identify and locate the
object), a media type attribute, a size attribute, an encoding object), a media type attribute, a size attribute, an encoding
attribute, or a message digest attribute. Since the set of objects attribute, or a message digest attribute. Since the set of objects
sent within a session can be dynamic, with new objects being added sent within a session can be dynamic, with new objects being added
and old ones removed, several instances of the FDT can be sent and a and old ones removed, several instances of the FDT can be sent and a
mechanism is provided to identify a new FDT Instance. mechanism is provided to identify a new FDT Instance.
Error detection and verification of the protocol control information
relies on the on the underlying transport (e.g., UDP checksum).
To provide robustness against packet loss and improve the efficiency To provide robustness against packet loss and improve the efficiency
of the on-demand mode, FLUTE/ALC relies on packet erasure coding (AL- of the on-demand mode, FLUTE/ALC relies on packet erasure coding (AL-
FEC). AL-FEC encoding is proactive (since there is no feedback and FEC). AL-FEC encoding is proactive (since there is no feedback and
therefore no (N)ACK-based retransmission) and ALC packets containing therefore no (N)ACK-based retransmission) and ALC packets containing
repair data are sent along with ALC packets containing source data. repair data are sent along with ALC packets containing source data.
Several FEC Schemes have been standardized; FLUTE/ALC does not Several FEC Schemes have been standardized; FLUTE/ALC does not
mandate the use of any particular one. Several strategies concerning mandate the use of any particular one. Several strategies concerning
the transmission order of ALC source and repair packets are possible, the transmission order of ALC source and repair packets are possible,
in particular in on-demand mode where it can deeply impact the in particular in on-demand mode where it can deeply impact the
service provided (e.g., to favor the recovery of objects in sequence, service provided (e.g., to favor the recovery of objects in sequence,
skipping to change at page 29, line 16 skipping to change at page 28, line 13
([RFC6726], section 1.1.4). FLUTE/ALC is often used over a network ([RFC6726], section 1.1.4). FLUTE/ALC is often used over a network
path with pre-provisioned capacity [I-D.ietf-tsvwg-rfc5405bis] where path with pre-provisioned capacity [I-D.ietf-tsvwg-rfc5405bis] where
there are no flows competing for capacity. In this case, a sender- there are no flows competing for capacity. In this case, a sender-
based rate control mechanism and a single channel is sufficient. based rate control mechanism and a single channel is sufficient.
[RFC6584] provides per-packet authentication, integrity, and anti- [RFC6584] provides per-packet authentication, integrity, and anti-
replay protection in the context of the ALC and NORM protocols. replay protection in the context of the ALC and NORM protocols.
Several mechanisms are proposed that seamlessly integrate into these Several mechanisms are proposed that seamlessly integrate into these
protocols using the ALC and NORM header extension mechanisms. protocols using the ALC and NORM header extension mechanisms.
4.9.2. Interface Description 3.9.2. Interface Description
The FLUTE/ALC specification does not describe a specific application The FLUTE/ALC specification does not describe a specific application
programming interface (API) to control protocol operation. programming interface (API) to control protocol operation.
Open source reference implementations of FLUTE/ALC are available at Open source reference implementations of FLUTE/ALC are available at
http://planete-bcast.inrialpes.fr/ (no longer maintained) and http://planete-bcast.inrialpes.fr/ (no longer maintained) and
http://mad.cs.tut.fi/ (no longer maintained), and these http://mad.cs.tut.fi/ (no longer maintained), and these
implementations specify and document their own APIs. Commercial implementations specify and document their own APIs. Commercial
versions are also available, some derived from the above versions are also available, some derived from the above
implementations, with their own API. implementations, with their own API.
4.9.3. Transport Features 3.9.3. Transport Features
The transport features provided by FLUTE/ALC are: The transport features provided by FLUTE/ALC are:
o unicast o unicast, multicast, anycast or IPv4 broadcast transmission,
o multicast, anycast or IPv4 broadcast.
o per-object dynamic meta-data delivery.
o push delivery or on-demand delivery service. o object-oriented delivery of discrete data or files and associated
metadata,
o fully reliable or partially reliable delivery (of file or in- o fully reliable or partially reliable delivery (of file or in-
memory objects). memory objects), using proactive packet erasure coding (AL-FEC) to
recover from packet erasures,
o ordered or unordered delivery (of file or in-memory objects). o ordered or unordered delivery (of file or in-memory objects),
o per-packet authentication, integrity, and anti-replay services. o error detection (based on the UDP checksum),
o proactive packet erasure coding (AL-FEC) to recover from packet o per-packet authentication,
erasures and improve the on-demand delivery service,
o error detection (through UDP). o per-packet integrity,
o per-packet replay protection,
o congestion control for layered flows (e.g., with WEBRC). o congestion control for layered flows (e.g., with WEBRC).
4.10. NACK-Oriented Reliable Multicast (NORM) 3.10. NACK-Oriented Reliable Multicast (NORM)
NORM is an IETF standards track protocol specified in [RFC5740]. The NORM is an IETF standards track protocol specified in [RFC5740]. It
protocol was designed to support reliable bulk data dissemination to provides object-oriented delivery of discrete data or files.
receiver groups using IP Multicast but also provides for point-to-
The protocol was designed to support reliable bulk data dissemination
to receiver groups using IP Multicast but also provides for point-to-
point unicast operation. Support for bulk data dissemination point unicast operation. Support for bulk data dissemination
includes discrete file or computer memory-based "objects" as well as includes discrete file or computer memory-based "objects" as well as
byte- and message-streaming. NORM is designed to incorporate packet byte- and message-streaming.
erasure coding as an inherent part of its selective ARQ in response
to receiver negative acknowledgments. The packet erasure coding can
also be proactively applied for forward protection from packet loss.
NORM transmissions are governed by the TCP-friendly congestion
control. NORM's reliability, congestion control, and flow control
mechanism are distinct components and can be separately controlled to
meet different application needs.
4.10.1. Protocol Description NORM can incorporate packet erasure coding as a part of its selective
ARQ in response to negative acknowledgments from the receiver. The
packet erasure coding can also be proactively applied for forward
protection from packet loss. NORM transmissions are governed by the
TCP-friendly congestion control. The reliability, congestion control
and flow control mechanisms can be separately controlled to meet
different application needs.
3.10.1. Protocol Description
The NORM protocol is encapsulated in UDP datagrams and thus provides The NORM protocol is encapsulated in UDP datagrams and thus provides
multiplexing for multiple sockets on hosts using port numbers. For multiplexing for multiple sockets on hosts using port numbers. For
loosely coordinated IP Multicast, NORM is not strictly connection- loosely coordinated IP Multicast, NORM is not strictly connection-
oriented although per-sender state is maintained by receivers for oriented although per-sender state is maintained by receivers for
protocol operation. [RFC5740] does not specify a handshake protocol protocol operation. [RFC5740] does not specify a handshake protocol
for connection establishment and separate session initiation can be for connection establishment. Separate session initiation can be
used to coordinate port numbers. However, in-band "client-server" used to coordinate port numbers. However, in-band "client-server"
style connection establishment can be accomplished with the NORM style connection establishment can be accomplished with the NORM
congestion control signaling messages using port binding techniques congestion control signaling messages using port binding techniques
like those for TCP client-server connections. like those for TCP client-server connections.
NORM supports bulk "objects" such as file or in-memory content but NORM supports bulk "objects" such as file or in-memory content but
also can treat a stream of data as a logical bulk object for purposes also can treat a stream of data as a logical bulk object for purposes
of packet erasure coding. In the case of stream transport, NORM can of packet erasure coding. In the case of stream transport, NORM can
support either byte streams or message streams where application- support either byte streams or message streams where application-
defined message boundary information is carried in the NORM protocol defined message boundary information is carried in the NORM protocol
messages. This allows the receiver(s) to join/re- join and recover messages. This allows the receiver(s) to join/re-join and recover
message boundaries mid-stream as needed. Application content is message boundaries mid-stream as needed. Application content is
carried and identified by the NORM protocol with encoding symbol carried and identified by the NORM protocol with encoding symbol
identifiers depending upon the Forward Error Correction (FEC) Scheme identifiers depending upon the Forward Error Correction (FEC) Scheme
[RFC3452] configured. NORM uses NACK-based selective ARQ to reliably [RFC3452] configured. NORM uses NACK-based selective ARQ to reliably
deliver the application content to the receiver(s). NORM proactively deliver the application content to the receiver(s). NORM proactively
measures round- trip timing information to scale ARQ timers measures round-trip timing information to scale ARQ timers
appropriately and to support congestion control. For multicast appropriately and to support congestion control. For multicast
operation, timer-based feedback suppression is uses to achieve group operation, timer-based feedback suppression is uses to achieve group
size scaling with low feedback traffic levels. The feedback size scaling with low feedback traffic levels. The feedback
suppression is not applied for unicast operation. suppression is not applied for unicast operation.
NORM uses rate-based congestion control based upon the TCP-Friendly NORM uses rate-based congestion control based upon the TCP-Friendly
Rate Control (TFRC) [RFC4324] principles that are also used in DCCP Rate Control (TFRC) [RFC4324] principles that are also used in DCCP
[RFC4340]. NORM uses control messages to measure RTT and collect [RFC4340]. NORM uses control messages to measure RTT and collect
congestion event (e..g, loss event, ECN event, etc) information from congestion event information (e.g., reflecting a loss event or ECN
the receiver(s) to support dynamic rate control adjustment. The TCP- event) from the receiver(s) to support dynamic adjustment or the
Friendly Multicast Congestion Control (TFMCC) [RFC4654] used provides rate. The TCP-Friendly Multicast Congestion Control (TFMCC)
some extra features to support multicast but is functionally [RFC4654] provides extra features to support multicast, but is
equivalent to TFRC in the unicast case. functionally equivalent to TFRC for unicast.
NORM's reliability mechanism is decoupled from congestion control. Error detection and verification of the protocol control information
This allows alternative arrangements of transport services to be relies on the on the underlying transport(e.g., UDP checksum).
invoked. For example, fixed-rate reliable delivery can be supported
or unreliable (but optionally "better than best effort" via packet
erasure coding) delivery with rate- control per TFRC can be achieved.
Additionally, alternative congestion control techniques may be
applied. For example, TFRC rate control with congestion event
detection based on ECN for links with high packet loss (e.g.,
wireless) has been implemented and demonstrated with NORM.
While NORM is NACK-based for reliability transfer, it also supports a The reliability mechanism is decoupled from congestion control. This
allows invocation of alternative arrangements of transport services.
For example, to support, fixed-rate reliable delivery or unreliable
delivery (that may optionally be "better than best effort" via packet
erasure coding) using TFRC. Alternative congestion control
techniques may be applied. For example, TFRC rate control with
congestion event detection based on ECN.
While NORM provides NACK-based reliability, it also supports a
positive acknowledgment (ACK) mechanism that can be used for receiver positive acknowledgment (ACK) mechanism that can be used for receiver
flow control. Again, since this mechanism is decoupled from the flow control. This mechanism is decoupled from the reliability and
reliability and congestion control, applications that have different congestion control, supporting applications with different needs.
needs in this aspect can use the protocol differently. One example One example is use of NORM for quasi-reliable delivery, where timely
is the use of NORM for quasi-reliable delivery where timely delivery delivery of newer content may be favored over completely reliable
of newer content may be favored over completely reliable delivery of delivery of older content within buffering and RTT constraints.
older content within buffering and RTT constraints.
4.10.2. Interface Description 3.10.2. Interface Description
The NORM specification does not describe a specific application The NORM specification does not describe a specific application
programming interface (API) to control protocol operation. A freely- programming interface (API) to control protocol operation. A freely-
available, open source reference implementation of NORM is available available, open source reference implementation of NORM is available
at https://www.nrl.navy.mil/itd/ncs/products/norm, and a documented at https://www.nrl.navy.mil/itd/ncs/products/norm, and a documented
API is provided for this implementation. While a sockets-like API is API is provided for this implementation. While a sockets-like API is
not currently documented, the existing API supports the necessary not currently documented, the existing API supports the necessary
functions for that to be implemented. functions for that to be implemented.
4.10.3. Transport Features 3.10.3. Transport Features
The transport features provided by NORM are: The transport features provided by NORM are:
o unicast or multicast transport. o unicast or multicast transport,
o stream-oriented delivery in a single stream.
o object-oriented delivery of discrete data or file items. o unidirectional communication,
o reliable delivery. o stream-oriented delivery in a single stream or object-oriented
delivery of in-memory data or file bulk content objects,
o unordered unidirectional delivery (of in-memory data or file bulk o fully reliable (NACK-based) or partially reliable (using erasure
content objects). coding both proactively and as part of ARQ) delivery,
o error detection (UDP checksum). o unordered delivery,
o segmentation. o error detection (relies on UDP checksum),
o data bundling (Nagle's algorithm). o segmentation,
o flow control (timer-based and/or ack-based). o data bundling (using Nagle's algorithm),
o congestion control. o flow control (timer-based and/or ack-based),
o packet erasure coding (both proactively and as part of ARQ). o congestion control (also supporting fixed rate reliable or
unreliable delivery).
4.11. Transport Layer Security (TLS) and Datagram TLS (DTLS) as a 3.11. Transport Layer Security (TLS) and Datagram TLS (DTLS) as a
pseudotransport pseudotransport
Transport Layer Security (TLS) and Datagram TLS (DTLS) are IETF Transport Layer Security (TLS) [RFC5246]} and Datagram TLS (DTLS)
protocols that provide several security-related features to [RFC6347]} are IETF protocols that provide several security-related
applications. TLS is designed to run on top of a reliable streaming features to applications. TLS is designed to run on top of a
transport protocol (usually TCP), while DTLS is designed to run on reliable streaming transport protocol (usually TCP), while DTLS is
top of a best-effort datagram protocol (UDP or DCCP [RFC5238]). At designed to run on top of a best-effort datagram protocol (UDP or
the time of writing, the current version of TLS is 1.2; which is DCCP [RFC5238]). At the time of writing, the current version of TLS
defined in [RFC5246]. DTLS provides nearly identical functionality is 1.2; which is defined in [RFC5246]. DTLS provides nearly
to applications; it is defined in [RFC6347] and its current version identical functionality to applications; it is defined in [RFC6347]
is also 1.2. The TLS protocol evolved from the Secure Sockets Layer and its current version is also 1.2. The TLS protocol evolved from
(SSL) protocols developed in the mid 90s to support protection of the Secure Sockets Layer (SSL) protocols developed in the mid-1990s
HTTP traffic. to support protection of HTTP traffic.
While older versions of TLS and DTLS are still in use, they provide While older versions of TLS and DTLS are still in use, they provide
weaker security guarantees. [RFC7457] outlines important attacks on weaker security guarantees. [RFC7457] outlines important attacks on
TLS and DTLS. [RFC7525] is a Best Current Practices (BCP) document TLS and DTLS. [RFC7525] is a Best Current Practices (BCP) document
that describes secure configurations for TLS and DTLS to counter that describes secure configurations for TLS and DTLS to counter
these attacks. The recommendations are applicable for the vast these attacks. The recommendations are applicable for the vast
majority of use cases. majority of use cases.
4.11.1. Protocol Description 3.11.1. Protocol Description
Both TLS and DTLS provide the same security features and can thus be Both TLS and DTLS provide the same security features and can thus be
discussed together. The features they provide are: discussed together. The features they provide are:
o Confidentiality o Confidentiality
o Data integrity o Data integrity
o Peer authentication (optional) o Peer authentication (optional)
o Perfect forward secrecy (optional) o Perfect forward secrecy (optional)
The authentication of the peer entity can be omitted; a common web The authentication of the peer entity can be omitted; a common web
use case is where the server is authenticated and the client is not. use case is where the server is authenticated and the client is not.
TLS also provides a completely anonymous operation mode in which TLS also provides a completely anonymous operation mode in which
neither peer's identity is authenticated. It is important to note neither peer's identity is authenticated. It is important to note
that TLS itself does not specify how a peering entity's identity that TLS itself does not specify how a peering entity's identity
should be interpreted. For example, in the common use case of should be interpreted. For example, in the common use case of
authentication by means of an X.509 certificate, it is the authentication by means of an X.509 certificate, it is the
application's decision whether the certificate of the peering entity application's decision whether the certificate of the peering entity
is acceptable for authorization decisions. Perfect forward secrecy, is acceptable for authorization decisions.
if enabled and supported by the selected algorithms, ensures that
traffic encrypted and captured during a session at time t0 cannot be Perfect forward secrecy, if enabled and supported by the selected
later decrypted at time t1 (t1 > t0), even if the long-term secrets algorithms, ensures that traffic encrypted and captured during a
of the communicating peers are later compromised. session at time t0 cannot be later decrypted at time t1 (t1 > t0),
even if the long-term secrets of the communicating peers are later
compromised.
As DTLS is generally used over an unreliable datagram transport such As DTLS is generally used over an unreliable datagram transport such
as UDP, applications will need to tolerate lost, re-ordered, or as UDP, applications will need to tolerate lost, re-ordered, or
duplicated datagrams. Like TLS, DTLS conveys application data in a duplicated datagrams. Like TLS, DTLS conveys application data in a
sequence of independent records. However, because records are mapped sequence of independent records. However, because records are mapped
to unreliable datagrams, there are several features unique to DTLS to unreliable datagrams, there are several features unique to DTLS
that are not applicable to TLS: that are not applicable to TLS:
o Record replay detection (optional). o Record replay detection (optional).
skipping to change at page 33, line 45 skipping to change at page 32, line 47
Generally, DTLS follows the TLS design as closely as possible. To Generally, DTLS follows the TLS design as closely as possible. To
operate over datagrams, DTLS includes a sequence number and limited operate over datagrams, DTLS includes a sequence number and limited
forms of retransmission and fragmentation for its internal forms of retransmission and fragmentation for its internal
operations. The sequence number may be used for detecting replayed operations. The sequence number may be used for detecting replayed
information, according to the windowing procedure described in information, according to the windowing procedure described in
Section 4.1.2.6 of [RFC6347]. DTLS forbids the use of stream Section 4.1.2.6 of [RFC6347]. DTLS forbids the use of stream
ciphers, which are essentially incompatible when operating on ciphers, which are essentially incompatible when operating on
independent encrypted records. independent encrypted records.
4.11.2. Interface Description 3.11.2. Interface Description
TLS is commonly invoked using an API provided by packages such as TLS is commonly invoked using an API provided by packages such as
OpenSSL, wolfSSL, or GnuTLS. Using such APIs entails the OpenSSL, wolfSSL, or GnuTLS. Using such APIs entails the
manipulation of several important abstractions, which fall into the manipulation of several important abstractions, which fall into the
following categories: long-term keys and algorithms, session state, following categories: long-term keys and algorithms, session state,
and communications/connections. There may also be special APIs and communications/connections. There may also be special APIs
required to deal with time and/or random numbers, both of which are required to deal with time and/or random numbers, both of which are
needed by a variety of encryption algorithms and protocols. needed by a variety of encryption algorithms and protocols.
Considerable care is required in the use of TLS APIs to ensure Considerable care is required in the use of TLS APIs to ensure
skipping to change at page 34, line 24 skipping to change at page 33, line 26
As an example, in the case of OpenSSL, the primary abstractions are As an example, in the case of OpenSSL, the primary abstractions are
the library itself and method (protocol), session, context, cipher the library itself and method (protocol), session, context, cipher
and connection. After initializing the library and setting the and connection. After initializing the library and setting the
method, a cipher suite is chosen and used to configure a context method, a cipher suite is chosen and used to configure a context
object. Session objects may then be minted according to the object. Session objects may then be minted according to the
parameters present in a context object and associated with individual parameters present in a context object and associated with individual
connections. Depending on how precisely the programmer wishes to connections. Depending on how precisely the programmer wishes to
select different algorithmic or protocol options, various levels of select different algorithmic or protocol options, various levels of
details may be required. details may be required.
4.11.3. Transport Features 3.11.3. Transport Features
Both TLS and DTLS employ a layered architecture. The lower layer is Both TLS and DTLS employ a layered architecture. The lower layer is
commonly called the record protocol. It is responsible for: commonly called the record protocol. It is responsible for:
o message fragmentation. o message fragmentation,
o authentication and integrity via message authentication codes o authentication and integrity via message authentication codes
(MAC). (MAC),
o data encryption. o data encryption,
o scheduling transmission using the underlying transport protocol. o scheduling transmission using the underlying transport protocol.
DTLS augments the TLS record protocol with: DTLS augments the TLS record protocol with:
o ordering and replay protection, implemented using sequence o ordering and replay protection, implemented using sequence
numbers. numbers.
Several protocols are layered on top of the record protocol. These Several protocols are layered on top of the record protocol. These
include the handshake, alert, and change cipher spec protocols. include the handshake, alert, and change cipher spec protocols.
skipping to change at page 35, line 9 skipping to change at page 34, line 11
compression parameters when a connection is first set up. In DTLS, compression parameters when a connection is first set up. In DTLS,
this protocol also has a basic fragmentation and retransmission this protocol also has a basic fragmentation and retransmission
capability and a cookie-like mechanism to resist DoS attacks. (TLS capability and a cookie-like mechanism to resist DoS attacks. (TLS
compression is not recommended at present). The alert protocol is compression is not recommended at present). The alert protocol is
used to inform the peer of various conditions, most of which are used to inform the peer of various conditions, most of which are
terminal for the connection. The change cipher spec protocol is used terminal for the connection. The change cipher spec protocol is used
to synchronize changes in cryptographic parameters for each peer. to synchronize changes in cryptographic parameters for each peer.
The data protocol, when used with an appropriate cipher, provides: The data protocol, when used with an appropriate cipher, provides:
o authentication of one end or both ends of a connection. o authentication of one end or both ends of a connection,
o confidentiality. o confidentiality,
o cryptographic integrity protection. o cryptographic integrity protection.
4.12. Hypertext Transport Protocol (HTTP) over TCP as a pseudotransport Both TLS and DTLS are unicast-only.
3.12. Hypertext Transport Protocol (HTTP) over TCP as a pseudotransport
The Hypertext Transfer Protocol (HTTP) is an application-level The Hypertext Transfer Protocol (HTTP) is an application-level
protocol widely used on the Internet. Version 1.1 of the protocol is protocol widely used on the Internet. It provides object-oriented
delivery of discrete data or files. Version 1.1 of the protocol is
specified in [RFC7230] [RFC7231] [RFC7232] [RFC7233] [RFC7234] specified in [RFC7230] [RFC7231] [RFC7232] [RFC7233] [RFC7234]
[RFC7235], and version 2 in [RFC7540]. HTTP is usually transported [RFC7235], and version 2 in [RFC7540]. HTTP is usually transported
over TCP using port 80 and 443, although it can be used with other over TCP using port 80 and 443, although it can be used with other
transports. When used over TCP it inherits its properties. transports. When used over TCP it inherits its properties.
HTTP is used as a substrate for other application-layer protocols. Application layer protocols may use HTTP as a substrate with an
There are various reasons for this practice listed in [RFC3205]; existing method and data formats, or specify new methods and data
these include being a well-known and well-understood protocol, formats. There are various reasons for this practice listed in
reusability of existing servers and client libraries, easy use of [RFC3205]; these include being a well-known and well-understood
existing security mechanisms such as HTTP digest authentication protocol, reusability of existing servers and client libraries, easy
[RFC2617] and TLS [RFC5246], the ability of HTTP to traverse use of existing security mechanisms such as HTTP digest
firewalls makes it work over many types of infrastructure, and in authentication [RFC2617] and TLS [RFC5246], the ability of HTTP to
cases where a application server often needs to support HTTP anyway. traverse firewalls makes it work over many types of infrastructure,
and in cases where an application server often needs to support HTTP
anyway.
Depending on application need, the use of HTTP as a substrate Depending on application need, the use of HTTP as a substrate
protocol may add complexity and overhead in comparison to a special- protocol may add complexity and overhead in comparison to a special-
purpose protocol (e.g., HTTP headers, suitability of the HTTP purpose protocol (e.g., HTTP headers, suitability of the HTTP
security model, etc.). [RFC3205] addresses this issue and provides security model, etc.). [RFC3205] addresses this issue and provides
some guidelines and concerns about the use of HTTP standard port 80 some guidelines and identifies concerns about the use of HTTP
and 443, the use of HTTP URL scheme and interaction with existing standard port 80 and 443, the use of HTTP URL scheme and interaction
firewalls, proxies and NATs. with existing firewalls, proxies and NATs.
4.12.1. Protocol Description Representational State Transfer (REST) [REST] is another example of
how applications can use HTTP as transport protocol. REST is an
architecture style that may be used to build applications using HTTP
as a communication protocol.
3.12.1. Protocol Description
Hypertext Transfer Protocol (HTTP) is a request/response protocol. A Hypertext Transfer Protocol (HTTP) is a request/response protocol. A
client sends a request containing a request method, URI and protocol client sends a request containing a request method, URI and protocol
version followed by a MIME-like message (see [RFC7231] for the version followed by a MIME-like message (see [RFC7231] for the
differences between an HTTP object and a MIME message), containing differences between an HTTP object and a MIME message), containing
information about the client and request modifiers. The message can information about the client and request modifiers. The message can
contain a message body carrying application data as well. The server also contain a message body carrying application data. The server
responds with a status or error code followed by a MIME-like message responds with a status or error code followed by a MIME-like message
containing information about the server and information about carried containing information about the server and information about the
data and it can include a message body. It is possible to specify a data. This may include a message body. It is possible to specify a
data format for the message body using MIME media types [RFC2045]. data format for the message body using MIME media types [RFC2045].
Furthermore, the protocol has numerous additional features; features The protocol has additional features, some relevant to pseudo-
relevant to pseudotransport are described below. transport are described below.
Content negotiation, specified in [RFC7231], is a mechanism provided Content negotiation, specified in [RFC7231], is a mechanism provided
by HTTP for selecting a representation on a requested resource. The by HTTP to allow selection of a representation for a requested
client and server negotiate acceptable data formats, charsets, data resource. The client and server negotiate acceptable data formats,
encoding (e.g., data can be transferred compressed using gzip), etc. character sets, data encoding (e.g., data can be transferred
HTTP can accommodate exchange of messages as well as data streaming compressed using gzip). HTTP can accommodate exchange of messages as
(using chunked transfer encoding [RFC7230]). It is also possible to well as data streaming (using chunked transfer encoding [RFC7230]).
request a part of a resource using range requests specified in It is also possible to request a part of a resource using an object
[RFC7233]. The protocol provides powerful cache control signalling range request [RFC7233]. The protocol provides powerful cache
defined in [RFC7234]. control signaling defined in [RFC7234].
HTTP 1.1's and HTTP 2.0's persistent connections can be use to The persistent connections of HTTP 1.1 and HTTP 2.0 allow multiple
perform multiple request-response transactions during the life-time request- response transactions (streams) during the life-time of a
of a single HTTP connection. Moreover, HTTP 2.0 connections can single HTTP connection. HTTP 2.0 connections can multiplex many
multiplex many request/response pairs in parallel on a single request/response pairs in parallel on a single transport connection.
transport connection. This reduces connection establishment overhead This reduces overhead during connection establishment and mitigates
and the effect of the transport layer slow-start on each transaction, transport layer slow-start that would have otherwise been incurred
important in reducing latency for HTTP's primary use case. for each transaction. Both are important to reduce latency for
HTTP's primary use case.
It is possible to combine HTTP with security mechanisms, like TLS HTTP can be combined with security mechanisms, such as TLS (denoted
(denoted by HTTPS), which adds protocol properties provided by such a by HTTPS). This adds protocol properties provided by such a
mechanism (e.g., authentication, encryption). The TLS Application- mechanism (e.g., authentication, encryption). The TLS Application-
Layer Protocol Negotiation (ALPN) extension [RFC7301] can be used for Layer Protocol Negotiation (ALPN) extension [RFC7301] can be used to
HTTP version negotiation within the TLS handshake, which eliminates negotiate the HTTP version within the TLS handshake, eliminating the
the latency of addition round-trips. Arbitrary cookie strings, latency incurred by additional round-trip exchanges. Arbitrary
included as part of the MIME headers, are often used as bearer tokens cookie strings, included as part of the MIME headers, are often used
in HTTP. as bearer tokens in HTTP.
Application layer protocols using HTTP as substrate may use an
existing method and data formats, or specify new methods and data
formats. Furthermore some protocols may not fit a request/response
paradigm and instead rely on HTTP to send messages (e.g., [RFC6546]).
Because HTTP works in many restricted infrastructures, it is also
used to tunnel other application-layer protocols.
4.12.2. Interface Description 3.12.2. Interface Description
There are many HTTP libraries available exposing different APIs. The There are many HTTP libraries available exposing different APIs. The
APIs provide a way to specify a request by providing a URI, a method, APIs provide a way to specify a request by providing a URI, a method,
request modifiers and optionally a request body. For the response, request modifiers and optionally a request body. For the response,
callbacks can be registered that will be invoked when the response is callbacks can be registered that will be invoked when the response is
received. If TLS is used, API expose a registration of callbacks in received. If TLS is used, the API exposes a registration of
case a server requests client authentication and when certificate callbacks for a server that requests client authentication and when
verification is needed. certificate verification is needed.
World Wide Web Consortium (W3C) standardized the XMLHttpRequest API The World Wide Web Consortium (W3C) has standardized the
[XHR], an API that can be use for sending HTTP/HTTPS requests and XMLHttpRequest API [XHR]. This API can be used for sending HTTP/
receiving server responses. Besides XML data format, request and HTTPS requests and receiving server responses. Besides the XML data
response data format can also be JSON, HTML and plain text. format, the request and response data format can also be JSON, HTML,
Specifically JavaScript and XMLHttpRequest are a ubiquitous and plain text. JavaScript and XMLHttpRequest are ubiquitous
programming model for websites, and more general applications, where programming models for websites, and more general applications, where
native code is less attractive. native code is less attractive.
Representational State Transfer (REST) [REST] is another example how 3.12.3. Transport features
applications can use HTTP as transport protocol. REST is an
architecture style for building application on the Internet. It uses
HTTP as a communication protocol.
4.12.3. Transport features The transport features provided by HTTP, when used as a pseudo-
transport, are:
The transport features provided by HTTP, when used as a o unicast transport (provided by the lower layer protocol, usually
pseudotransport, are: TCP),
o unicast. o uni- or bidirectional communication,
o message and stream-oriented transfer. o transfer of objects in multiple streams with object content type
negotiation, supporting partial transmission of object ranges,
o bi- or unidirectional transmission. o ordered delivery (provided by the lower layer protocol, usually
TCP),
o ordered delivery. o fully reliable delivery (provided by the lower layer protocol,
usually TCP),
o fully reliable delivery. o flow control (provided by the lower layer protocol, usually TCP).
o object range request. o congestion control (provided by the lower layer protocol, usually
TCP).
o message content type negotiation. HTTPS (HTTP over TLS) additionally provides the following features
(as provided by TLS):
o flow control. o authentication (of one or both ends of a connection),
HTTPS (HTTP over TLS) additionally provides the following components: o confidentiality,
o authentication (of one or both ends of a connection). o integrity protection.
o confidentiality. 4. Congestion Control
o integrity protection. Congestion control is critical to the stable operation of the
Internet. A variety of mechanisms are used to provide the congestion
control needed by many Internet transport protocols. Congestion is
detected based on sensing of network conditions, whether through
explicit or implicit feedback. The congestion control mechanisms
that can be applied by different transport protocols are largely
orthogonal to the choice of transport protocol. This section
provides an overview of the congestion control mechanisms available
to the protocols described in Section 3.
5. Transport Service Features Many protocols use a separate window to determine the maximum sending
rate that is allowed by the congestion control. The used congestion
control mechanism will increase the congestion window if feedback is
received that indicates that the currently used network path is not
congested, and will reduce the window otherwise. Window-based
mechanisms often increase their window slowing over multiple RTTs,
while decreasing strongly when the first indication of congestion is
received. One example are Additive Increase Multiplicative Decrease
(AIMD) schemes, where the window is increased by a certain number of
packets/bytes for each data segment that has been successfully
transmitted, while the window is multiplicatively decrease on the
occurrence of a congestion event. This can lead to a rather
unstable, oscillating sending rate, but will resolve a congestion
situation quickly. TCP New Reno [RFC5681] which is one of the
initial proposed schemes for TCP as well as TCP Cubic
[I-D.ietf-tcpm-cubic] which is the default mechanism for TCP in Linux
are two examples for window-based AIMD schemes. This approach is
also used by DCCP CCID-2 for datagram congestion control.
Some classes of applications prefer to use a transport service that
allows sending at a more stable rate, that is slowly varied in
response to congestion. Rate-based methods offer this type of
congestion control and have been defined based on the loss ratio and
observed round trip time, such as TFRC [RFC5348] and TFRC-SP
[RFC4828]. These methods utilize a throughput equation to determine
the maximum acceptable rate. Such methods are used with DCCP CCID-3
[RFC4342] and CCID-4 [RFC5622], WEBRC [RFC3738], and other
applications.
Another class of applications prefer a transport service that yields
to other (higher-priority) traffic, such as interactive
transmissions. While most traffic in the Internet uses loss-based
congestion control and therefore need to fill the network buffers (to
a certain level if Active Queue Management (AQM) is used), low-
priority congestion control methods often react to changes in delay
as an earlier indication of congestion. This approach tends to
induce less loss than a loss-based method but does generally not
compete well with loss-based traffic across shared bottleneck links.
Therefore, methods such as LEDBAT [RFC6824], are deployed in the
Internet for scavenger traffic that aim to only utilize otherwise
unused capacity.
5. Transport Features
The tables below summarize some key features to illustrate the range The tables below summarize some key features to illustrate the range
of functions provided across the IETF-specified transports. Figure 1 of functions provided across the IETF-specified transports. Figure 1
considers transports that may be directly layered over the network, considers transports that may be directly layered over the network,
and Figure 2 considers transports layered over another transport and Figure 2 considers transports layered over another transport
service. service. Features that are permitted, but not required, are marked
as "Poss" indicating that it is possible for the transport service to
offer this feature.
+---------------+------+------+------+------+------+------+------+ +---------------+------+------+------+------+------+------+------+
| Feature | TCP | MPTCP| SCTP | UDP | UDP-L|DCCP |ICMP | | Feature | TCP | MPTCP| SCTP | UDP | UDP-L|DCCP |ICMP |
+---------------+------+------+------+------+------+------+------+ +---------------+------+------+------+------+------+------+------+
| Datagram | No | No | Yes | Yes | Yes | Yes | Yes | | Datagram | No | No | Yes | Yes | Yes | Yes | Yes |
+---------------+------+------+------+------+------+------+------+ +---------------+------+------+------+------+------+------+------+
| Conn. Oriented| Yes | Yes | Yes | No | No | Yes | No | | Conn. Oriented| Yes | Yes | Yes | No | No | Yes | No |
+---------------+------+------+------+------+------+------+------+ +---------------+------+------+------+------+------+------+------+
| Reliability | Yes | Yes | Yes | No | No | No | No | | Reliability | Yes | Yes | Yes | No | No | No | No |
+---------------+------+------+------+------+------+------+------+ +---------------+------+------+------+------+------+------+------+
| Partial Rel. | No | No | Pos | N/A | N/A | Yes | N/A | | Partial Rel. | No | No | Poss | N/A | N/A | Yes | N/A |
+---------------+------+------+------+------+------+------+------+ +---------------+------+------+------+------+------+------+------+
| Corupt. Tol | No | No | No | No | Yes | Yes | No | | Corupt. Tol | No | No | No | No | Yes | Yes | No |
+---------------+------+------+------+------+------+------+------+ +---------------+------+------+------+------+------+------+------+
| Cong.Control | Yes | Yes | Yes | No | No | Yes | No | | Cong.Control | Yes | Yes | Yes | No | No | Yes | No |
+---------------+------+------+------+------+------+------+------+ +---------------+------+------+------+------+------+------+------+
| Endpoint | 1 | >=1 | >=1 | 1 | 1 | 1 | 1 | | Endpoint | 1 | >=1 | >=1 | 1 | 1 | 1 | 1 |
+---------------+------+------+------+------+------+------+------+ +---------------+------+------+------+------+------+------+------+
| Multicast Cap.| No | No | No | Yes | Yes | No | No | | Multicast Cap.| No | No | No | Yes | Yes | No | No |
+---------------+------+------+------+------+------+------+------+ +---------------+------+------+------+------+------+------+------+
Figure 1: Summary comparison: Transport protocols Figure 1: Summary comparison: Transport protocols
+---------------+------+------+------+------+------+ +---------------+------+------+------+------+------+
| Feature | RTP | FLUTE| NORM |(D)TLS| HTTP | | Feature | RTP | FLUTE| NORM |(D)TLS| HTTP |
+---------------+------+------+------+------+------+ +---------------+------+------+------+------+------+
| Datagram | Yes | No | Both | Both | No | | Datagram | Yes | No | Both | Both | No |
+---------------+------+------+------+------+------+ +---------------+------+------+------+------+------+
| Conn. Oriented| No | Yes | Yes | Yes | Yes | | Conn. Oriented| No | Yes | Yes | Yes | Yes |
+---------------+------+------+------+------+------+ +---------------+------+------+------+------+------+
| Reliability | No | Yes | Pos | Pos | Yes | | Reliability | No | Yes | Poss | Poss | Yes |
+---------------+------+------+------+------+------+ +---------------+------+------+------+------+------+
| Partial R | Pos | No | Pos | No | No | | Partial R | Poss | No | Poss | No | No |
+---------------+------+------+------+------+------+ +---------------+------+------+------+------+------+
| Corupt. Tol | Poss | No | No | No | No | | Corupt. Tol | Poss | No | No | No | No |
+---------------+------+------+------+------+------+ +---------------+------+------+------+------+------+
| Cong.Control | Poss | Poss | Poss | N/A | N/A | | Cong.Control | Poss | Poss | Poss | N/A | N/A |
+---------------+------+------+------+------+------+ +---------------+------+------+------+------+------+
| Endpoint | >=1 | >=1 | >=1 | 1 | 1 | | Endpoint | >=1 | >=1 | >=1 | 1 | 1 |
+---------------+------+------+------+------+------+ +---------------+------+------+------+------+------+
| Multicast Cap.| Yes | Yes | Yes | No | No | | Multicast Cap.| Yes | Yes | Yes | No | No |
+---------------+------+------+------+------+------+ +---------------+------+------+------+------+------+
Figure 2: Upper layer transports and frameworks Figure 2: Upper layer transports and frameworks
The transport protocol components analyzed in this document that can The transport protocol features described in this document could be
be used as a basis for defining common transport service features, used as a basis for defining common transport features:
normalized and separated into categories, are as follows:
o Control Functions o Control Functions
* Addressing * Addressing
+ unicast (TCP, MPTCP, SCTP, UDP, UDP-Lite, DCCP, TLS, HTTP) + unicast (TCP, MPTCP, SCTP, UDP, UDP-Lite, DCCP, ICMP, RTP,
TLS, HTTP)
+ multicast (UDP, UDP-Lite, DCCP, FLUTE/ALC, NORM) + multicast (UDP, UDP-Lite, DCCP, ICMP, RTP, FLUTE/ALC, NORM).
Note that, as TLS and DTLS are unicast-only, there is no
widely deployed mechanism for supporting the features in the
Security section below when using multicast addressing.
+ IPv4 broadcast (UDP, UDP-Lite, DCCP) + IPv4 broadcast (UDP, UDP-Lite, ICMP)
+ anycast (UDP, UDP-Lite, DCCP). Connection-oriented + anycast (UDP, UDP-Lite). Connection-oriented protocols such
protocols such as TCP can be and are used with anycast as TCP and DCCP have also been deployed using anycast
routing, with the risk that routing changes may cause addressing, with the risk that routing changes may cause
connection failure. connection failure.
* Association type
+ connection-oriented (TCP, MPTCP, SCTP, DCCP, RTP, NORM, TLS,
HTTP)
+ connectionless (UDP, UDP-Lite, FLUTE/ALC)
* Multihoming support * Multihoming support
+ multihoming for resilience (MPTCP, SCTP) + resilience and mobility (MPTCP, SCTP)
+ multihoming for mobility (MPTCP, SCTP) + load-balancing (MPTCP)
+ multihoming for load-balancing (MPTCP) + address family multiplexing (MPTCP, SCTP)
* Application to port mapping (TCP, MPTCP, SCTP, UDP, UDP-Lite, * Middlebox cooperation
DCCP, FLUTE/ALC, NORM, TLS, HTTP)
+ with commonly deployed support in NAPT (TCP, MPTCP, UDP, + application-class signaling to middleboxes (DCCP)
TLS, HTTP)
+ error condition signaling from middleboxes and routers to
endpoints (ICMP)
* Signaling
+ control information and error signaling (ICMP)
+ performance metric reporting (RTP)
o Delivery o Delivery
* reliability * Reliability
+ fully reliable delivery (TCP, MPTCP, SCTP, FLUTE/ALC, NORM, + fully reliable delivery (TCP, MPTCP, SCTP, FLUTE/ALC, NORM,
TLS, HTTP) TLS, HTTP)
+ partially reliable delivery (SCTP, NORM) + partially reliable delivery (SCTP, NORM)
- using packet erasure coding (NORM, FLUTE, RTP) - using packet erasure coding (FLUTE/ALC, NORM, RTP)
+ unreliable delivery (SCTP, UDP, UDP-Lite, DCCP) - with specified policy for dropped messages (SCTP)
- with drop notification (SCTP, DCCP) + unreliable delivery (SCTP, UDP, UDP-Lite, DCCP, RTP)
+ Integrity protection - with drop notification to sender (RTP, SCTP, DCCP)
+ error detection
- checksum for error detection (TCP, MPTCP, SCTP, UDP, UDP- - checksum for error detection (TCP, MPTCP, SCTP, UDP, UDP-
Lite, DCCP, FLUTE/ALC, NORM, TLS, HTTP) Lite, DCCP, ICMP, FLUTE/ALC, NORM, TLS, DTLS)
- partial payload checksum protection (UDP-Lite, DCCP) - partial payload checksum protection (UDP-Lite, DCCP).
Some uses of RTP can exploit partial payload checksum
protection feature to provide a corruption tolerant
transport service.
- checksum optional (UDP) - checksum optional (UDP). Possible with IPv4 and in
certain cases with IPv6.
* ordering * Ordering
+ ordered delivery (TCP, MPTCP, SCTP, TLS, HTTP) + ordered delivery (TCP, MPTCP, SCTP, RTP, FLUTE, TLS, HTTP)
+ unordered delivery (SCTP, UDP, UDP-Lite, DCCP, NORM) + unordered delivery permitted (SCTP, UDP, UDP-Lite, DCCP,
RTP, NORM)
* type/framing * Type/framing
+ stream-oriented delivery (TCP, MPTCP, SCTP, TLS) + stream-oriented delivery (TCP, MPTCP, SCTP, TLS, HTTP)
- with multiple streams per association (SCTP) - with multiple streams per association (SCTP, HTTP2)
+ message-oriented delivery (SCTP, UDP, UDP-Lite, DCCP, DTLS) + message-oriented delivery (SCTP, UDP, UDP-Lite, DCCP, RTP,
DTLS)
+ object-oriented delivery of discrete data or file items + object-oriented delivery of discrete data or files and
(FLUTE/ALC, NORM, HTTP) associated metadata (FLUTE/ALC, NORM, HTTP)
- with partial delivery of object ranges (HTTP)
* Directionality
+ unidirectional (TCP, SCTP, UDP, UDP-Lite DCCP, RTP, FLUTE/
ALC, NORM)
+ bidirectional (TCP, MPTCP, SCTP, HTTP, TLS)
o Transmission control o Transmission control
* flow control (TCP, MPTCP, SCTP, DCCP, TLS, HTTP) * flow control (TCP, MPTCP, SCTP, DCCP, RTP, TLS, HTTP)
* congestion control (TCP, MPTCP, SCTP, DCCP, FLUTE/ALC, NORM, * congestion control (TCP, MPTCP, SCTP, DCCP, RTP, FLUTE/ALC,
TLS, HTTP) NORM). Congestion control can also provided by the transport
supporting an upper later transport (e.g., RTP,HTTP, TLS).
* segmentation (TCP, MPTCP, SCTP, FLUTE/ALC, NORM, TLS, HTTP) * segmentation (TCP, MPTCP, SCTP, RTP, FLUTE/ALC, NORM, TLS,
HTTP)
* data/message bundling (TCP, MPTCP, SCTP, TLS, HTTP) * data/message bundling (TCP, MPTCP, SCTP, TLS, HTTP)
* stream scheduling prioritization (SCTP, HTTP2)
* stream scheduling prioritization (SCTP) * endpoint multiplexing (MPTCP)
o Security (may be used in combination with other transports) o Security
* authentication of one end of a connection (TLS) * authentication of one end of a connection (FLUTE/ALC, TLS,
DTLS)
* authentication of both ends of a connection (TLS) * authentication of both ends of a connection (TLS, DTLS)
* confidentiality (TLS) * confidentiality (TLS, DTLS)
* cryptographic integrity protection (TLS) * cryptographic integrity protection (TLS, DTLS)
* replay protection (FLUTE/ALC, DTLS)
6. IANA Considerations 6. IANA Considerations
This document has no considerations for IANA. This document has no considerations for IANA.
7. Security Considerations 7. Security Considerations
This document surveys existing transport protocols and protocols This document surveys existing transport protocols and protocols
providing transport-like services. Confidentiality, integrity, and providing transport-like services. Confidentiality, integrity, and
authenticity are among the features provided by those services. This authenticity are among the features provided by those services. This
document does not specify any new components or mechanisms for document does not specify any new features or mechanisms for
providing these features. Each RFC listed in this document discusses providing these features. Each RFC referenced by this document
the security considerations of the specification it contains. discusses the security considerations of the specification it
contains.
8. Contributors 8. Contributors
In addition to the editors, this document is the work of Brian In addition to the editors, this document is the work of Brian
Adamson, Dragana Damjanovic, Kevin Fall, Simone Ferlin-Oliviera, Adamson, Dragana Damjanovic, Kevin Fall, Simone Ferlin-Oliviera,
Ralph Holz, Olivier Mehani, Karen Nielsen, Colin Perkins, Vincent Ralph Holz, Olivier Mehani, Karen Nielsen, Colin Perkins, Vincent
Roca, and Michael Tuexen. Roca, and Michael Tuexen.
o Section 4.2 on MPTCP was contributed by Simone Ferlin-Oliviera o Section 3.2 on MPTCP was contributed by Simone Ferlin-Oliviera
(ferlin@simula.no) and Olivier Mehani (ferlin@simula.no) and Olivier Mehani
(olivier.mehani@nicta.com.au) (olivier.mehani@nicta.com.au)
o Section 4.4 on UDP was contributed by Kevin Fall (kfall@kfall.com) o Section 3.4 on UDP was contributed by Kevin Fall (kfall@kfall.com)
o Section 4.3 on SCTP was contributed by Michael Tuexen (tuexen@fh- o Section 3.3 on SCTP was contributed by Michael Tuexen (tuexen@fh-
muenster.de) and Karen Nielsen (karen.nielsen@tieto.com) muenster.de) and Karen Nielsen (karen.nielsen@tieto.com)
o Section 4.8 on RTP contains contributions from Colin Perlins o Section 3.8 on RTP contains contributions from Colin Perkins
(csp@csperkins.org) (csp@csperkins.org)
o Section 4.9 on FLUTE/ALC was contributed by Vincent Roca o Section 3.9 on FLUTE/ALC was contributed by Vincent Roca
(vincent.roca@inria.fr) (vincent.roca@inria.fr)
o Section 4.10 on NORM was contributed by Brian Adamson o Section 3.10 on NORM was contributed by Brian Adamson
(brian.adamson@nrl.navy.mil) (brian.adamson@nrl.navy.mil)
o Section 4.11 on TLS and DTLS was contributed by Ralph Holz o Section 3.11 on TLS and DTLS was contributed by Ralph Holz
(ralph.holz@nicta.com.au) and Olivier Mehani (ralph.holz@nicta.com.au) and Olivier Mehani
(olivier.mehani@nicta.com.au) (olivier.mehani@nicta.com.au)
o Section 4.12 on HTTP was contributed by Dragana Damjanovic o Section 3.12 on HTTP was contributed by Dragana Damjanovic
(ddamjanovic@mozilla.com) (ddamjanovic@mozilla.com)
9. Acknowledgments 9. Acknowledgments
Thanks to Joe Touch, Michael Welzl, and the TAPS Working Group for Thanks to Joe Touch, Michael Welzl, and the TAPS Working Group for
the comments, feedback, and discussion. This work is partially the comments, feedback, and discussion. This work is supported by
supported by the European Commission under grant agreements the European Commission under grant agreement No. 318627 mPlane and
FP7-ICT-318627 mPlane and from the Horizon 2020 research and from the Horizon 2020 research and innovation program under grant
innovation program under grant agreement No. 644334 (NEAT); support agreements No. 644334 (NEAT) and No. 688421 (MAMI). This support
does not imply endorsement. does not imply endorsement.
10. Informative References 10. Informative References
[RFC0768] Postel, J., "User Datagram Protocol", STD 6, RFC 768, [RFC0768] Postel, J., "User Datagram Protocol", STD 6, RFC 768,
DOI 10.17487/RFC0768, August 1980, DOI 10.17487/RFC0768, August 1980,
<http://www.rfc-editor.org/info/rfc768>. <http://www.rfc-editor.org/info/rfc768>.
[RFC0792] Postel, J., "Internet Control Message Protocol", STD 5, [RFC0792] Postel, J., "Internet Control Message Protocol", STD 5,
RFC 792, DOI 10.17487/RFC0792, September 1981, RFC 792, DOI 10.17487/RFC0792, September 1981,
skipping to change at page 43, line 44 skipping to change at page 45, line 14
[RFC3168] Ramakrishnan, K., Floyd, S., and D. Black, "The Addition [RFC3168] Ramakrishnan, K., Floyd, S., and D. Black, "The Addition
of Explicit Congestion Notification (ECN) to IP", of Explicit Congestion Notification (ECN) to IP",
RFC 3168, DOI 10.17487/RFC3168, September 2001, RFC 3168, DOI 10.17487/RFC3168, September 2001,
<http://www.rfc-editor.org/info/rfc3168>. <http://www.rfc-editor.org/info/rfc3168>.
[RFC3205] Moore, K., "On the use of HTTP as a Substrate", BCP 56, [RFC3205] Moore, K., "On the use of HTTP as a Substrate", BCP 56,
RFC 3205, DOI 10.17487/RFC3205, February 2002, RFC 3205, DOI 10.17487/RFC3205, February 2002,
<http://www.rfc-editor.org/info/rfc3205>. <http://www.rfc-editor.org/info/rfc3205>.
[RFC3260] Grossman, D., "New Terminology and Clarifications for
Diffserv", RFC 3260, DOI 10.17487/RFC3260, April 2002,
<http://www.rfc-editor.org/info/rfc3260>.
[RFC3436] Jungmaier, A., Rescorla, E., and M. Tuexen, "Transport [RFC3436] Jungmaier, A., Rescorla, E., and M. Tuexen, "Transport
Layer Security over Stream Control Transmission Protocol", Layer Security over Stream Control Transmission Protocol",
RFC 3436, DOI 10.17487/RFC3436, December 2002, RFC 3436, DOI 10.17487/RFC3436, December 2002,
<http://www.rfc-editor.org/info/rfc3436>. <http://www.rfc-editor.org/info/rfc3436>.
[RFC3450] Luby, M., Gemmell, J., Vicisano, L., Rizzo, L., and J. [RFC3450] Luby, M., Gemmell, J., Vicisano, L., Rizzo, L., and J.
Crowcroft, "Asynchronous Layered Coding (ALC) Protocol Crowcroft, "Asynchronous Layered Coding (ALC) Protocol
Instantiation", RFC 3450, DOI 10.17487/RFC3450, December Instantiation", RFC 3450, DOI 10.17487/RFC3450, December
2002, <http://www.rfc-editor.org/info/rfc3450>. 2002, <http://www.rfc-editor.org/info/rfc3450>.
skipping to change at page 45, line 26 skipping to change at page 46, line 45
Datagram Congestion Control Protocol (DCCP) Congestion Datagram Congestion Control Protocol (DCCP) Congestion
Control ID 3: TCP-Friendly Rate Control (TFRC)", RFC 4342, Control ID 3: TCP-Friendly Rate Control (TFRC)", RFC 4342,
DOI 10.17487/RFC4342, March 2006, DOI 10.17487/RFC4342, March 2006,
<http://www.rfc-editor.org/info/rfc4342>. <http://www.rfc-editor.org/info/rfc4342>.
[RFC4433] Kulkarni, M., Patel, A., and K. Leung, "Mobile IPv4 [RFC4433] Kulkarni, M., Patel, A., and K. Leung, "Mobile IPv4
Dynamic Home Agent (HA) Assignment", RFC 4433, Dynamic Home Agent (HA) Assignment", RFC 4433,
DOI 10.17487/RFC4433, March 2006, DOI 10.17487/RFC4433, March 2006,
<http://www.rfc-editor.org/info/rfc4433>. <http://www.rfc-editor.org/info/rfc4433>.
[RFC4614] Duke, M., Braden, R., Eddy, W., and E. Blanton, "A Roadmap
for Transmission Control Protocol (TCP) Specification
Documents", RFC 4614, DOI 10.17487/RFC4614, September
2006, <http://www.rfc-editor.org/info/rfc4614>.
[RFC4654] Widmer, J. and M. Handley, "TCP-Friendly Multicast [RFC4654] Widmer, J. and M. Handley, "TCP-Friendly Multicast
Congestion Control (TFMCC): Protocol Specification", Congestion Control (TFMCC): Protocol Specification",
RFC 4654, DOI 10.17487/RFC4654, August 2006, RFC 4654, DOI 10.17487/RFC4654, August 2006,
<http://www.rfc-editor.org/info/rfc4654>. <http://www.rfc-editor.org/info/rfc4654>.
[RFC4820] Tuexen, M., Stewart, R., and P. Lei, "Padding Chunk and [RFC4820] Tuexen, M., Stewart, R., and P. Lei, "Padding Chunk and
Parameter for the Stream Control Transmission Protocol Parameter for the Stream Control Transmission Protocol
(SCTP)", RFC 4820, DOI 10.17487/RFC4820, March 2007, (SCTP)", RFC 4820, DOI 10.17487/RFC4820, March 2007,
<http://www.rfc-editor.org/info/rfc4820>. <http://www.rfc-editor.org/info/rfc4820>.
skipping to change at page 50, line 35 skipping to change at page 51, line 45
[RFC7301] Friedl, S., Popov, A., Langley, A., and E. Stephan, [RFC7301] Friedl, S., Popov, A., Langley, A., and E. Stephan,
"Transport Layer Security (TLS) Application-Layer Protocol "Transport Layer Security (TLS) Application-Layer Protocol
Negotiation Extension", RFC 7301, DOI 10.17487/RFC7301, Negotiation Extension", RFC 7301, DOI 10.17487/RFC7301,
July 2014, <http://www.rfc-editor.org/info/rfc7301>. July 2014, <http://www.rfc-editor.org/info/rfc7301>.
[RFC7323] Borman, D., Braden, B., Jacobson, V., and R. [RFC7323] Borman, D., Braden, B., Jacobson, V., and R.
Scheffenegger, Ed., "TCP Extensions for High Performance", Scheffenegger, Ed., "TCP Extensions for High Performance",
RFC 7323, DOI 10.17487/RFC7323, September 2014, RFC 7323, DOI 10.17487/RFC7323, September 2014,
<http://www.rfc-editor.org/info/rfc7323>. <http://www.rfc-editor.org/info/rfc7323>.
[RFC7414] Duke, M., Braden, R., Eddy, W., Blanton, E., and A.
Zimmermann, "A Roadmap for Transmission Control Protocol
(TCP) Specification Documents", RFC 7414,
DOI 10.17487/RFC7414, February 2015,
<http://www.rfc-editor.org/info/rfc7414>.
[RFC7457] Sheffer, Y., Holz, R., and P. Saint-Andre, "Summarizing [RFC7457] Sheffer, Y., Holz, R., and P. Saint-Andre, "Summarizing
Known Attacks on Transport Layer Security (TLS) and Known Attacks on Transport Layer Security (TLS) and
Datagram TLS (DTLS)", RFC 7457, DOI 10.17487/RFC7457, Datagram TLS (DTLS)", RFC 7457, DOI 10.17487/RFC7457,
February 2015, <http://www.rfc-editor.org/info/rfc7457>. February 2015, <http://www.rfc-editor.org/info/rfc7457>.
[RFC7496] Tuexen, M., Seggelmann, R., Stewart, R., and S. Loreto, [RFC7496] Tuexen, M., Seggelmann, R., Stewart, R., and S. Loreto,
"Additional Policies for the Partially Reliable Stream "Additional Policies for the Partially Reliable Stream
Control Transmission Protocol Extension", RFC 7496, Control Transmission Protocol Extension", RFC 7496,
DOI 10.17487/RFC7496, April 2015, DOI 10.17487/RFC7496, April 2015,
<http://www.rfc-editor.org/info/rfc7496>. <http://www.rfc-editor.org/info/rfc7496>.
skipping to change at page 51, line 10 skipping to change at page 52, line 27
"Recommendations for Secure Use of Transport Layer "Recommendations for Secure Use of Transport Layer
Security (TLS) and Datagram Transport Layer Security Security (TLS) and Datagram Transport Layer Security
(DTLS)", BCP 195, RFC 7525, DOI 10.17487/RFC7525, May (DTLS)", BCP 195, RFC 7525, DOI 10.17487/RFC7525, May
2015, <http://www.rfc-editor.org/info/rfc7525>. 2015, <http://www.rfc-editor.org/info/rfc7525>.
[RFC7540] Belshe, M., Peon, R., and M. Thomson, Ed., "Hypertext [RFC7540] Belshe, M., Peon, R., and M. Thomson, Ed., "Hypertext
Transfer Protocol Version 2 (HTTP/2)", RFC 7540, Transfer Protocol Version 2 (HTTP/2)", RFC 7540,
DOI 10.17487/RFC7540, May 2015, DOI 10.17487/RFC7540, May 2015,
<http://www.rfc-editor.org/info/rfc7540>. <http://www.rfc-editor.org/info/rfc7540>.
[I-D.ietf-aqm-ecn-benefits]
Fairhurst, G. and M. Welzl, "The Benefits of using
Explicit Congestion Notification (ECN)", draft-ietf-aqm-
ecn-benefits-08 (work in progress), November 2015.
[I-D.ietf-tsvwg-rfc5405bis] [I-D.ietf-tsvwg-rfc5405bis]
Eggert, L., Fairhurst, G., and G. Shepherd, "UDP Usage Eggert, L., Fairhurst, G., and G. Shepherd, "UDP Usage
Guidelines", draft-ietf-tsvwg-rfc5405bis-07 (work in Guidelines", draft-ietf-tsvwg-rfc5405bis-07 (work in
progress), November 2015. progress), November 2015.
[I-D.ietf-aqm-ecn-benefits]
Fairhurst, G. and M. Welzl, "The Benefits of using
Explicit Congestion Notification (ECN)", draft-ietf-aqm-
ecn-benefits-07 (work in progress), November 2015.
[I-D.ietf-tsvwg-sctp-dtls-encaps] [I-D.ietf-tsvwg-sctp-dtls-encaps]
Tuexen, M., Stewart, R., Jesup, R., and S. Loreto, "DTLS Tuexen, M., Stewart, R., Jesup, R., and S. Loreto, "DTLS
Encapsulation of SCTP Packets", draft-ietf-tsvwg-sctp- Encapsulation of SCTP Packets", draft-ietf-tsvwg-sctp-
dtls-encaps-09 (work in progress), January 2015. dtls-encaps-09 (work in progress), January 2015.
[I-D.ietf-tsvwg-sctp-ndata] [I-D.ietf-tsvwg-sctp-ndata]
Stewart, R., Tuexen, M., Loreto, S., and R. Seggelmann, Stewart, R., Tuexen, M., Loreto, S., and R. Seggelmann,
"Stream Schedulers and User Message Interleaving for the "Stream Schedulers and User Message Interleaving for the
Stream Control Transmission Protocol", draft-ietf-tsvwg- Stream Control Transmission Protocol", draft-ietf-tsvwg-
sctp-ndata-04 (work in progress), July 2015. sctp-ndata-04 (work in progress), July 2015.
[I-D.ietf-tsvwg-sctp-failover] [I-D.ietf-tsvwg-sctp-failover]
Nishida, Y., Natarajan, P., Caro, A., Amer, P., and K. Nishida, Y., Natarajan, P., Caro, A., Amer, P., and K.
Nielsen, "SCTP-PF: Quick Failover Algorithm in SCTP", Nielsen, "SCTP-PF: Quick Failover Algorithm in SCTP",
draft-ietf-tsvwg-sctp-failover-13 (work in progress), draft-ietf-tsvwg-sctp-failover-14 (work in progress),
September 2015. December 2015.
[I-D.ietf-tsvwg-natsupp] [I-D.ietf-tsvwg-natsupp]
Stewart, R., Tuexen, M., and I. Ruengeler, "Stream Control Stewart, R., Tuexen, M., and I. Ruengeler, "Stream Control
Transmission Protocol (SCTP) Network Address Translation Transmission Protocol (SCTP) Network Address Translation
Support", draft-ietf-tsvwg-natsupp-08 (work in progress), Support", draft-ietf-tsvwg-natsupp-08 (work in progress),
July 2015. July 2015.
[I-D.ietf-tcpm-cubic]
Rhee, I., Xu, L., Ha, S., Zimmermann, A., Eggert, L., and
R. Scheffenegger, "CUBIC for Fast Long-Distance Networks",
draft-ietf-tcpm-cubic-00 (work in progress), June 2015.
[XHR] van Kesteren, A., Aubourg, J., Song, J., and H. Steen, [XHR] van Kesteren, A., Aubourg, J., Song, J., and H. Steen,
"XMLHttpRequest working draft "XMLHttpRequest working draft
(http://www.w3.org/TR/XMLHttpRequest/)", 2000. (http://www.w3.org/TR/XMLHttpRequest/)", 2000.
[REST] Fielding, R., "Architectural Styles and the Design of [REST] Fielding, R., "Architectural Styles and the Design of
Network-based Software Architectures, Ph. D. (UC Irvine), Network-based Software Architectures, Ph. D. (UC Irvine),
Chapter 5: Representational State Transfer", 2000. Chapter 5: Representational State Transfer", 2000.
[POSIX] 1-2008, IEEE., "IEEE Standard for Information Technology [POSIX] 1-2008, IEEE., "IEEE Standard for Information Technology
-- Portable Operating System Interface (POSIX) Base -- Portable Operating System Interface (POSIX) Base
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